Re: [Asterisk-Users] dropped calls from queue
I just updated to latest cvs and the problem remains. I did also notice that when the call coming in on the queue is through a Zap line (from an adtran 750 to an x100p) it logs the following just before the warnings below: pr 7 14:21:21 VERBOSE[60194841]: -- SIP/hrutter-432b answered Zap/13-1 Apr 7 14:21:21 DEBUG[60194841]: Set option TONE VERIFY, mode: MUTECONF/MAX(2) on Zap/13-1 Apr 7 14:21:21 VERBOSE[60194841]: -- Stopped music on hold on Zap/13-1 Tony Buser wrote: We're having a strange problem with our receptionist. She runs an xpro softphone and we're using a queue to handle incoming calls. It seems nearly all of the calls that come in through the queue get dropped. At first we thought it might have been human error (clicking the wrong button in xpro or something) or that the person waiting in the queue just gave up and hungup, however it seems to happen when the following gets logged: Apr 7 14:53:35 WARNING[60424217]: File 10 does not exist in any format Apr 7 14:53:35 WARNING[60424217]: Unable to open 10 (format G729A): No such file or directory Apr 7 14:53:35 WARNING[60424217]: Agent on SIP/hrutter-c6fa hungup on the customer. They're going to be pissed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dialpad.com
Greetings, Does anyone have any experience in getting dialpad.com working with * They use a proprietary softphone but also have facility for cisco ata-186 and Sipura SPA-2000. Before I go off and investigate, I though I would check and see if anyone has any experience with them Thanks, Craig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: pda skype
I'm going to leave most of what you said alone, I understand you point and it's your point to make. However I will make a small comment about “I don't need hotlist functionality, if I dial their number and they aren't on, I get a busy reorder signal. No big deal”. Presence based information is the biggest ‘seller’ in the IP PBX market at the moment, being able to tell what/where a person is certainly driving a lot of sales through my door. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Hall Sent: Thursday, 8 April 2004 3:29 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FW: pda skype I don't have a PocketPC PDA, mine is Palm. But regardless, I don't see what all the "Hype" is with "Skype." It is a closed protocol and highly platform-restricted product. Sure the concept of a peer-to-peer phone network is interesting, but if not everyone can connect to it, what is the point? If they want to keep certain features under their control so that they can eventually charge for it, then by all means do so. But if they would release the basic protocol specs, so that others can access the network in general, they would see many more users. I for one am not going to run yet another soft phone and/or IM client on my system just to connect to yet another phone network. A friend of mine and I tried it when it first came out, and it worked about as well as FWD, or IAXTel, or Firefly, or... You get the point. Now if I could attach my Asterisk server to it and be able to make and receive simple voice calls with other users, that would be great. I don't need hotlist functionality, if I dial their number and they aren't on, I get a busy reorder signal. No big deal. They definitely have a good idea, in the fact that it works, doesn't have too many problems with firewalls, and is not server reliant. But keeping it closed is preventing a lot of people from joining them.
Re: [Asterisk-Users] Getting info about changes in CVS
On Wed, 2004-04-07 at 17:20, Eric Wieling wrote: > There are several ways to know what changes in Asterisk's CVS. > This URL http://asterisk.gnuinter.net/files/changelogs/ contains fairly > up to date CVS changelog summary information. > You can also sign up for the Asterisk-CVS mailing list at > http://lists.digium.com/mailman/listinfo/asterisk-cvs > Archives of the Asterisk-CVS mailing list are at > http://lists.digium.com/pipermail/asterisk-cvs/ Any chance of adding this list to the GMane archive? For me browsing list archives via NNTP is *much* nicer than web interfaces... Thanks a lot, Fran. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Lucent Phones
Any phone you can plug into a regular POTS PSTN line from your Telco should work with the TDM400. Don't expect the fancy function buttons to work, however. On Wed, 2004-04-07 at 14:49, Gregory Junker wrote: > What about the Partner phones and TDM400? > > > You can't plug Lucent's (Avaya's) DCP, MLX, or ATL phone sets into an > > Asterisk box -- the protocols are all proprietary. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss." ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dropped calls from queue
Hate to reply to my own message again, but I just figured it out. Nothing wrong with asterisk really, just a bad configuration. Somehow the queue line in extensions conf got changed by someone to: exten => 81003,3,Queue(receptionistq|tTH||10) Thats where the 10 was coming from. :) Could this be considered a bug? It shouldn't hang up on someone just because the wav file for an announcement can't be found? All this time we were blaming the poor receptionist. Tony Buser wrote: Apr 7 14:53:35 WARNING[60424217]: File 10 does not exist in any format Apr 7 14:53:35 WARNING[60424217]: Unable to open 10 (format G729A): No such file or directory Apr 7 14:53:35 WARNING[60424217]: Agent on SIP/hrutter-c6fa hungup on the customer. They're going to be pissed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] error 488 - Not Acceptable Here
Take out the allow=all in your sip.conf and put in allow= for the codec
you want to use and disallow=all.
On Wed, 2004-04-07 at 15:18, Roger wrote:
> I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone
> are registered by the below information
>
> *CLI> sip show peers
> Name/usernameHost Mask Port Status
> 2002/2002192.168.22.199 (D) 255.255.255.255 5060 Unmonitored
> 2001/2001192.168.22.200 (D) 255.255.255.255 5060 Unmonitored
> 2000/2000192.168.22.198 (D) 255.255.255.255 5060 Unmonitored
>
> *CLI> sip show users
> Username Secret Authen Def.Context A/C
> 2002 ciscomd5,plaintextdemo No
> 2001 ciscomd5,plaintextdemo No
> 2000 ciscomd5,plaintextdemo No
>
>
> All 3 phones and the asterisk box are on the 192.168.22.0/24 subnet.
> I've attached my sip.conf and extensions.conf file for review...
>
> When I start the server and a phone dials another phone I get the below
> answer.
>
> *CLI> -- Executing Dial("SIP/2001-0bb5", "SIP/2002|30|tr") in new stack
> -- Called 2002
> -- Got SIP response 488 "Not Acceptable Here" back from 192.168.22.199
> == No one is available to answer at this time
> -- Timeout on SIP/2001-0bb5
>
> I *believe* the sip response might be from the phone itself - and not a
> asterisk misconfig. I'm just wanting a second pair of eyes.
>
> I put in
>
> canreinvite=no
>
> for each phone profile as people have said this is needed for buggy
> Cisco phones.
>
>
> __
> ;
> ; SIP Configuration for Asterisk
> ;
> ; Syntax for specifying a SIP device in extensions.conf is
> ; SIP/devicename where devicename is defined in a section below.
> ;
> ; You may also use
> ; SIP/[EMAIL PROTECTED] to call any SIP user on the Internet
> ; (Don't forget to enable DNS SRV records if you want to use this)
> ;
> ; If you define a SIP proxy as a peer below, you may call
> ; SIP/proxyhostname/user or SIP/[EMAIL PROTECTED]
> ; where the proxyhostname is defined in a section below
> ;
> ; Useful CLI commands to check peers/users:
> ; sip show peers Show all SIP peers (including friends)
> ; sip show users Show all SIP users (including friends)
> ; sip show registry Show status of hosts we register with
> ;
> ; sip debug Show all SIP messages
> ;
>
> [general]
>
> port = 5060 ; Port to bind to (SIP is 5060)
> bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
> ;bindaddr = 192.168.22.254; Address to bind to (all addresses on machine)
> allow=all ; Allow all codecs
> context = bogon-calls ; Send SIP callers that we don't know about here
> tos = lowdelay ; can be lowdelay, throughput, reliability, mincost
>
> [2000]
> type=friend ; This device takes and makes calls
> username=2000 ; Username on device
> secret=cisco ; Password for device
> ;host=192.168.22.1 ; This host is not on the same IP addr every time
> host=dynamic
> context=demo ; Inbound calls from this host go here
> mailbox=100 ; Activate the message waiting light if this
> canreinvite=no
> ; voicemailbox has messages in it
>
> [2001]; Duplicate of 2000, except with different auth data
> type=friend
> username=2001
> secret=cisco
> host=dynamic
> ;host=192.168.22.2
> context=demo
> mailbox=101
> canreinvite=no
>
> [2002]; Duplicate of 2000, except with different auth data
> type=friend
> username=2002
> secret=cisco
> ;host=192.168.22.3
> host=dynamic
> context=demo
> mailbox=102
> canreinvite=no
>
> __
> ;
> ; Static extension configuration file, used by
> ; the pbx_config module. This is where you configure all your
> ; inbound and outbound calls in Asterisk.
> ;
> [incoming]
> exten => s,1,Echo ;for testing the connection
> ;exten => s,1,Playback,demo-thanks ;for playing a file
> ;
> ; The "General" category is for certain variables.
> ;
> [general]
> ;
> ; If static is set to no, or omitted, then the pbx_config will rewrite
> ; this file when extensions are modified. Remember that all comments
> ; made in the file will be lost when that happens.
> ;
> ; XXX Not yet implemented XXX
> ;
> static=yes
> ;
> ; if static=yes and writeprotect=no, you can save dialplan by
> ; CLI command 'save dialplan' too
> ;
> writeprotect=no
>
> ; You can include other config files, use the #include command (without the ';')
> ; Note that this is different from the "include" command that includes contexts
> within
> ; other contexts. The #include command works in all asterisk configuration files.
> ;#include "filename.conf"
RE: [Asterisk-Users] Channel Bank?
I'd take a look at the VoiceTronic cards ( http://www.voicetronix.com/hda.htm ) which can be used with * or their free software.. these cards can be configured as : 12 Loop-Start ports only. 8 Loop-Start AND 4 Station ports. 4 Loop-Start AND 8 Station ports (default configuration). 12 Station ports only. HTH Andy *** REPLY SEPARATOR *** On 07/04/2004 at 08:00 John Vogel wrote: >Four or five analog lines can be done with a single computer so no channel >bank is needed. If you need 6 or more than there is also the choice of >using >two machines and IAX. > >-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] On Behalf Of Robert Hajime >Lanning >Sent: Tuesday, April 06, 2004 12:01 AM >To: [EMAIL PROTECTED] >Subject: Re: [Asterisk-Users] Channel Bank? > > > >> Hello, I'm new to Asterisk and would like to know how you could have 4 >> to 6 incoming analog POTS lines connecting to the Asterisk server and >> have 4 to 6 analog lines going out.(A T1 line is too costly). Would 2 >> channel banks be used? > >A T1 channelbank has 24 channels, so only 1 is needed. > >FXO channels (What you connect to the POTS lines) can be both inbound and >outbound. If you are not using DID. So, you just need to find out how >many >concurrent calls you need to support. > >If you are using analog DID lines, then those are inbound only, and require >FXS ports. (You supply dialtone and battery, the carrier's switch picks up >your line and dials into your PBX.) > >Now, there are multiple ways to get the analog lines into Asterisk... > o use an external gateway... POTS <-> SIP <-> Asterisk > o wait until next month and get the FXO multiport cards from Digium > o get a T1 card + channelbank > >-- >END OF LINE > -MCP >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question
If you change the extension to the follow
exten => *55,1,VoiceMailMain(${CALLERIDNUM})
The voicemail will now user their caller id for the mailbox
>
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[Asterisk-Users] inband dtmfmode, SIP to VoicePulse, > 1 digit extentions do not work?
I have a situation where calls come in via SIP from VoicePulse and get dropped into a main menu.Voicepulse only works /w dtmfmode=inband and I only allow ulaw/alaw as codecs. When the call comes in and gets dropped to the menu, you can hit 1, 2 or 3 to get to other people. Or you can dial the persons full extention... 1000 1001 etc... /w dtmfmode=inband, it only ever grabs the first digit rather than collecting "tones" till it hits the DigitTimeout. I tried from a SIP hardphone that uses rfc2833 and it worked as expected (collected digits until the timeout, then tried routing). Anyone know WHY this happens and if there is some sort of workaround? Barring that, does anyone know a SIP or IAX provider that will give me inbound/outbound calling /w a 212 areacode phone number ( ie + 1 212 - XXX- ) and uses dtmfmode = info or rfc2833 ? Thanks a bunch everyone, -jwb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question
This is a fairly simple thing to do. You don;t say what type of phones you are using,
so I;ll assume SIP for the example:
Step 1:
Put
callerid="Darren" <1234>
for each phone definition in sip.conf, obviously replacing Darren with the user eg
"Darren Nay" or "Joe Bloggs", then replace the 1234 with their extension number. The
format looks a little odd with the <1234> outside the quotes, but it's correct.
In your extensions.conf:
exten => *55,1,Ringing
exten => *55,2,Wait(1) ; Make the user feel like something is happening
exten => *55,3,VoiceMailMain(s${CALLERIDNUM})
The last line will not prompt for a password, if you omit the 's' it will...
HTH
Andy
*** REPLY SEPARATOR ***
On 07/04/2004 at 16:23 Darren Nay wrote:
>Hey All,
>
>
>
>We are using Asterisks as a voicemail only application, and so far all is
>great. (Excellent product!)
>
>
>
>However, I do have one question that I am hoping you might be able to help
>me with.
>
>
>
>In our asterisk application. When our customers call *55 (our dialplan
>code
>to check voicemail) then they are sent directly to voicemail (asterisk).
>Asterisk then gives a voice prompt asking the customer to enter their
>extension number (entire 10 digit telephone number in our case).
>
>
>
>My question is. Is there a way to make asterisk aware of the calling-from
>(callerID) number so that it will automatically detect the number and then
>go directly to asking them to input their password.
>
>
>
>If so, where would I make the config changes for this in the asterisk
>config
>files, and does anyone have an example of a similar config?
>
>
>
>Thanks!
>
>
>
>Darren Nay
>
>VOIP Network Developer
>
>Ionosphere, Inc
>
>[EMAIL PROTECTED]
>
>
>
>
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Re: [Asterisk-Users] res_motv: Request for Comment
On Tue, 6 Apr 2004, Mark Spencer wrote: > I've been considering the nature of Asterisk, its security, the bug > tracker, and more... And i've come up with an interesting idea: A > "message of the version". The idea is that Asterisk has a compile time > 32-bit unsigned int version which is incremented whenever some major new > bug is fixed. When Asterisk starts up (and periodically, maybe once per > day), it sends a packet with the version number to a server at Digium, > along with a message level (INFO,MINOR,MAJOR,CRITICAL) and the Digium > server replies (if it receives the packet, if not, it might get sent again > in a day) with any INFO, MINOR, MAJOR, or CRITICAL messages which are > associated with that version of the code. In this way, an asterisk > administrator could easily see if there were any major issues, critical > security updates, etc, that his system might need to be updated for. This could easily be done with simple dns lookups and TXT records, eg do a TXT query for version#.digium.com. The nice thing is that because of the distributed and cached nature of DNS, it is inherently resistant to high loads and outages -- especially if you have secondary/tertiary servers. -Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lucent Phones
This may be a little to far into PBX land but... Anyone know of a place where there are good examples of how to configure the Definity PBX stations with PRI? I currently have a T1 between a Definity and Asterisk. It is currently doing robbed-bit signalling but I would like to do PRI. I can get the PRI up and working signalling wise but to actually route calls from the PBX to Asterisk is another story. More specifically I am trying to figure out how to do this: PSTN->PBX->Asterisk->IP Phone I need to get an extension on the PBX to leave out the T1 to the Asterisk box. Thanks, Steve Matthew Branton wrote: Absolutely, it can be a little tricky but its definitely doable. Check out the info I wrote on the wiki, as well as peoples posts here for more information on hows its done. Matt -Original Message- From: James Moran [mailto:[EMAIL PROTECTED] Sent: Wednesday, April 07, 2004 1:39 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Lucent Phones Does Asterisk work with Lucent or any other PBX phone systems ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: pda skype
I guess I didn’t place that part of my message in the correct context. Presence is very handy, and I would like to see the functionality added to Asterisk. What I meant by my comment you quoted below, is that if I could attach my Asterisk server to the Skype network, I would not care if I had their presence services. That is a part of the functionality I would be willing to do without if they opened their protocol. It goes back to my original point: If they want to keep part of their protocol closed, that is fine, but release the most basic part of it: the voice communication. Jeremy From: Dean Collins [mailto:[EMAIL PROTECTED] Sent: Wednesday, April 07, 2004 2:42 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FW: pda skype I'm going to leave most of what you said alone, I understand you point and it's your point to make. However I will make a small comment about “I don't need hotlist functionality, if I dial their number and they aren't on, I get a busy reorder signal. No big deal”. Presence based information is the biggest ‘seller’ in the IP PBX market at the moment, being able to tell what/where a person is certainly driving a lot of sales through my door. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Hall Sent: Thursday, 8 April 2004 3:29 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FW: pda skype I don't have a PocketPC PDA, mine is Palm. But regardless, I don't see what all the "Hype" is with "Skype." It is a closed protocol and highly platform-restricted product. Sure the concept of a peer-to-peer phone network is interesting, but if not everyone can connect to it, what is the point? If they want to keep certain features under their control so that they can eventually charge for it, then by all means do so. But if they would release the basic protocol specs, so that others can access the network in general, they would see many more users. I for one am not going to run yet another soft phone and/or IM client on my system just to connect to yet another phone network. A friend of mine and I tried it when it first came out, and it worked about as well as FWD, or IAXTel, or Firefly, or... You get the point. Now if I could attach my Asterisk server to it and be able to make and receive simple voice calls with other users, that would be great. I don't need hotlist functionality, if I dial their number and they aren't on, I get a busy reorder signal. No big deal. They definitely have a good idea, in the fact that it works, doesn't have too many problems with firewalls, and is not server reliant. But keeping it closed is preventing a lot of people from joining them.
[Asterisk-Users] H.323 Seg faulting
Can someone take a look, tell me if this is a bug, a possible resources issue, or my own damn fault? http://bugs.digium.com/bug_view_page.php?bug_id=0001381 Thanks, Derek
[Asterisk-Users] Presence (was FW: pda skype)
Dean Collins just sent out a message a second ago (responding to an earlier posting regarding the new Skype PDA client). He said: "Presence based information is the biggest 'seller' in the IP PBX market at the moment, being able to tell what/where a person is certainly driving a lot of sales through my door." I would like to take a moment to second his message. Every presentation at the VON show concurs with his opinion. PRESENCE IS LIKELY THE MOST IMPORTANT SINGLE ADDITION TO ASTERISK THAT COULD BE MADE AT THIS TIME. I know there are people out there working on or waiting for all kinds of features. And I understand that all of them are important in one way or another. But THE feature that turns up time and again on RFPs for VoIP phone systems is PRESENCE. Like it or not, managers like to know where their people are. Friends like knowing where their friends are. Everybody likes being able to communicate how they want, when they want. This is the next step beyond the follow-me/find me applications. Beyond the basics of VoIP. And this step is driving people to deploy VoIP systems (including Asterisk). Perhaps the gory details of implementation are best reserved for the Developer list, but I think everybody out there can comment on the ideas: HOW DO YOU SEE PRESENCE INTEGRATING WITH ASTERISK? Thanks, Steve Steven Sokol Owner/Manager Sokol & Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web:http://www.sokol-associates.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Lucent Phones
Also, check out www.citel.com This company claims to have SIP adaptors for Avaya's digital PBX phones. If they work as advertised, you can keep your Avaya/Lucent phones, throw out your legacy PBX, and connect them all to Asterisk! However, I doubt they have all the display integration working correctly. Anyone know for sure? Ron Dutt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Sokol Sent: Wednesday, April 07, 2004 1:25 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Lucent Phones > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > Subject: [Asterisk-Users] Lucent Phones > > Does Asterisk work with Lucent or any other PBX phone systems > Sure. You can use Asterisk as a VoIP gateway to your existing legacy PBX. You can't plug Lucent's (Avaya's) DCP, MLX, or ATL phone sets into an Asterisk box -- the protocols are all proprietary. But you can certainly connect between the systems using analog or T1/Ei connections. Regards, Steve Steven Sokol Owner/Manager Sokol & Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web:http://www.sokol-associates.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Lucent Phones
On Wed, 2004-04-07 at 15:44 -0500, Eric Wieling wrote: > Don't expect the fancy function buttons to > work, however. > That's specifically what I was asking about... Has anyone tried to decipher the ETR signaling protocol? Or is it such a closely guarded Lucent/Avaya secret as to make the formula for Coca-Cola look like an open-source recipe? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question
On Apr 7, 2004, at 4:23 PM, Darren Nay wrote:
My question is. Is there a way to make asterisk aware of the
calling-from (callerID) number so that it will automatically detect
the number and then go directly to asking them to input their
password.
From "show application VoicemailMain" try:
exten => 1001,1,VoiceMailMain(${CALLERIDNUM})
We use:
exten => 1001,1,VoiceMailMain(s${CALLERIDNUM})
to skip password
Change extensions accordingly.
Jeb
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[Asterisk-Users] Problems with ADIT 600 - latency, loss, etc
I'm emailing this as the customer in this case, since my carrier appears to be completely unable to solve this. A brief rundown of the problem: - We have several voice lines going through the ADIT, of course into a VoIP type of arrangement. - Voice traffic will become choppy, even drop calls completely, at random but quite often. Modem calls (which we needed unfortunately) are a joke. - Packet latency on the data side will be fine one minute, then high the next. As I type this email through an ssh session into my server there, I hit keys and wait for them to appear. This is odd for a T1 in my opinion. Now, the carrier has replaced the hardware (breaking it twice in the process, leaving me to wonder how much of this they understand) and installed the latest hardware revision supposedly. They are claiming that high data usage is causing the voice issue. I don't buy it, and here's why: - The ADIT is designed to handle both and prioritize voice over data. This therefore has to use the assumption that no matter what, data will attempt to run at full speed, and the ADIT will throttle it back as necessary. The carrier claims that using data at high speed will overload the CPU causing dropped calls. Computers connect at 100mbit ethernet - unsure what he wants me to do here. I call BS on this one, and in fact did, with silence as the response. - I have enabled packet queueing in my firewall, with a max outbound badnwidth of the T1 speed. This enables me to also monitor my realtime usage in 5 second polls. While pushing out around 300kbit (on a T1), the voice call I was on was choppy, and I had packet latency. 1.5Mbit or 300kbit seems to yield the same results. - It happens consistently, even in low-usage times. My websites are dead at midnight, but phone calls still suck. - My firewall now prioritizes all of my ssh traffic above anything else, meaning that even if I was pushing out full speed ahead, that traffic gets top priority - so, there should be no delay whatsoever. I have suggested a check of the physical line. The technicians coming out only seem to be able to console into the ADIT and break it, and apparently our idea that a simple BER test on the line might enlighten them has not been welcomed. Am I off-base here? Is there some setting or other check that can be done in the ADIT? I'd like to see the physical T1 validated but am becoming increasingly concerned that no one at this company knows how to do it (I'm hoping I don't need to bring in a Tberd and have them loop it, so I can do their job for them). I found something on here about the T1 slipping and/or getting bipolar violations. What else can I have these guys check? Thanks - Ralph Rather annoyed ADIT user who thinks it's perhaps not the box's fault... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: pda skype
It cant be that hard to do considering Siemens are offering a cordless handset that can connect to skype. I guess it’s just a matter of bridging the 2 together. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Hall Sent: Thursday, 8 April 2004 7:20 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FW: pda skype I guess I didn’t place that part of my message in the correct context. Presence is very handy, and I would like to see the functionality added to Asterisk. What I meant by my comment you quoted below, is that if I could attach my Asterisk server to the Skype network, I would not care if I had their presence services. That is a part of the functionality I would be willing to do without if they opened their protocol. It goes back to my original point: If they want to keep part of their protocol closed, that is fine, but release the most basic part of it: the voice communication. Jeremy From: Dean Collins [mailto:[EMAIL PROTECTED] Sent: Wednesday, April 07, 2004 2:42 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FW: pda skype I'm going to leave most of what you said alone, I understand you point and it's your point to make. However I will make a small comment about “I don't need hotlist functionality, if I dial their number and they aren't on, I get a busy reorder signal. No big deal”. Presence based information is the biggest ‘seller’ in the IP PBX market at the moment, being able to tell what/where a person is certainly driving a lot of sales through my door. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Hall Sent: Thursday, 8 April 2004 3:29 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FW: pda skype I don't have a PocketPC PDA, mine is Palm. But regardless, I don't see what all the "Hype" is with "Skype." It is a closed protocol and highly platform-restricted product. Sure the concept of a peer-to-peer phone network is interesting, but if not everyone can connect to it, what is the point? If they want to keep certain features under their control so that they can eventually charge for it, then by all means do so. But if they would release the basic protocol specs, so that others can access the network in general, they would see many more users. I for one am not going to run yet another soft phone and/or IM client on my system just to connect to yet another phone network. A friend of mine and I tried it when it first came out, and it worked about as well as FWD, or IAXTel, or Firefly, or... You get the point. Now if I could attach my Asterisk server to it and be able to make and receive simple voice calls with other users, that would be great. I don't need hotlist functionality, if I dial their number and they aren't on, I get a busy reorder signal. No big deal. They definitely have a good idea, in the fact that it works, doesn't have too many problems with firewalls, and is not server reliant. But keeping it closed is preventing a lot of people from joining them.
RE: [Asterisk-Users] WAMi - Windows Asterisk Manager
Doesn't work for me. Connects to Asterisk but says "All extensions are busy right now" when I try to do anything. Here's what an extension looks like. Any suggestions? Thanks! Ext 2003 2003 SIP/2003 from-sip -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick Sent: Tuesday, April 06, 2004 5:46 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] WAMi - Windows Asterisk Manager I've got this one running on two different systems hitting two different Asterisk boxes, one a home and one at work. I like the looks, but I'm having problems being able to transfer calls using the 'Drag & Drop' or the button method. Sometimes it works and sometimes it coughs up an error. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Hoffmeyer Sent: Monday, April 05, 2004 11:36 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] WAMi - Windows Asterisk Manager Thank you for all of the beta testing. New and improved graphics in this release along with drag and drop transfers and hold for all technologies. There's a screenshot on the link below. Also improved documentation so read the included README. There's also a sample xml configuration included. http://www.voip-info.org/tiki-index.php?page=Asterisk+WAMI Christian Hoffmeyer YottaDot Solutions Huntsville, AL (iax) 700.859.4508 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.645 / Virus Database: 413 - Release Date: 3/28/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.645 / Virus Database: 413 - Release Date: 3/28/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Lucent Phones
> On Wed, 2004-04-07 at 15:44 -0500, Eric Wieling wrote: > > Don't expect the fancy function buttons to > > work, however. > > > > That's specifically what I was asking about... > > Has anyone tried to decipher the ETR signaling protocol? Or is it such a > closely guarded Lucent/Avaya secret as to make the formula for Coca-Cola > look like an open-source recipe? > ETR (Enhanced Tip/Ring) supposedly uses some variety of serial protocol over two lines to provide the screen functionality. The voice channel is still POTS. These phones are sold with the Partner system and can be added to the Magix systems using an ETR blade. Here in the states we could be jailed for trying to reverse engineer the serial display protocol (at least, in theory -- I don't know if it's been tested yet) as a violation of the evil DCMA. Anybody know about the other core Avaya protocols: specifically DCP and its cousin TDL. DCP (Digital Communications Protocol) has been used for years on the Definity line. Somewhere in the mid 1990s they cut it from 4 wires to two wires. The two wire version was ported to the Merlin Magix platform and is called "TDL" which I have been told means "Two wire DigitaL"?? If you have a large investment in the Avaya sets, it might be nice to have a bridge device to convert to SIP or H323 or whatever. I saw somebody at VON who offered a device that they claimed did just that. Don't remember who just now. Regs, -S ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] error 488 - Not Acceptable Here
Eric Wieling wrote: Take out the allow=all in your sip.conf and put in allow= for the codec you want to use and disallow=all. Holy crap it worked! sip.conf disallow=all ; disallow all codecs allow=ulaw ; Allow all codecs allow=alaw ; Allow all codecs ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Lucent Phones
Right, I know that the voice part is POTS because I have a standard cordless phone plugged into our Partner system. Hmm, wouldn't ETR be covered under a patent and not a copyright? And has 17 years been up yet? And if someone is selling devices that convert to/from ETR, then the protocol spec is available in some form (even if it's some draconian Avaya licensing scheme). I agree that Avaya has a vested interest in keeping the spec out of the public eye (sell phone upgrades, sell Merlin adapter modules), but this technology is definitely getting long in the toothwhich doesn't mean that my users exactly want to give up the familiarity of the Partner phones just yet. ;) And since I already have Partner phones, and don't really care to spend $200-$300 a pop to replace them with Snom or Cisco phones (good as they may be)... My goal is to get rid of that box on my wall. I already got rid of one (Cisco 1720 that was our router, replaced by a Linux server/router), now I have two to go (Lucent ConnectReach for our Time Warner Telecom IBL, and the Partner ACS phone system). Hell, Lucent Technologies ought to pay me rent for the amount of space they occupy on my walls. [rant=on] It is completely obnoxious to me that I have to take an incoming channelized T1 and have it broken out into physical copper wire so that I can insert it into my Partner system for voice. If I had then to take that copper, spend beaucoup more bucks to be able to put it back INTO digital form so that it can work with an Asterisk PBX...that's borderline surreal to me. Everyone is so vested in making sure that none of their damned equipment interoperates with anyone else's (yet all the while paying serious lip service to the holy grail of "standards") that I am to the point where DCMA be damned, if I can measure it I can figure it out. It pisses me off no end that TWTC can't simply send a normal T1 into my business (and therefore allow me to use a simple T100P), and I'll bet that when they start offering VoIP in this area (SW Ohio) it'll also involve some absurd piece of proprietary equipment further to clutter up my wall or rack. [rant=off] At any rate, yes, I could pick up a TDM400 and have Asterisk act like Partner ACS analog extensions, or pick up 3 X100's and use it directly for the incoming lines (and then deal with the user fallout regarding adaptation to X-Lite or something similar), but I just can't bring myself to do it, honestly. Ultimately, I want those boxes off my wall because technologically, they do not need to be there. Guess I'm stuck with finding 7960's on eBay as cheap as I can. *sigh* Anyone want an outmoded Partner ACS R1.0 analog phone system? ;) Greg On Wed, 2004-04-07 at 17:46 -0500, Steven Sokol wrote: > > On Wed, 2004-04-07 at 15:44 -0500, Eric Wieling wrote: > > > Don't expect the fancy function buttons to > > > work, however. > > > > > > > That's specifically what I was asking about... > > > > Has anyone tried to decipher the ETR signaling protocol? Or is it such a > > closely guarded Lucent/Avaya secret as to make the formula for Coca-Cola > > look like an open-source recipe? > > > > ETR (Enhanced Tip/Ring) supposedly uses some variety of serial protocol over > two lines to provide the screen functionality. The voice channel is still > POTS. These phones are sold with the Partner system and can be added to the > Magix systems using an ETR blade. > > Here in the states we could be jailed for trying to reverse engineer the > serial display protocol (at least, in theory -- I don't know if it's been > tested yet) as a violation of the evil DCMA. > > Anybody know about the other core Avaya protocols: specifically DCP and its > cousin TDL. DCP (Digital Communications Protocol) has been used for years > on the Definity line. Somewhere in the mid 1990s they cut it from 4 wires > to two wires. The two wire version was ported to the Merlin Magix platform > and is called "TDL" which I have been told means "Two wire DigitaL"?? > > If you have a large investment in the Avaya sets, it might be nice to have a > bridge device to convert to SIP or H323 or whatever. I saw somebody at VON > who offered a device that they claimed did just that. Don't remember who > just now. > > Regs, > > -S > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Siemens EWSD 13
Hi, I had exactly the same symptoms today with a co-located * connected to a Public Switch here in the UK. The problem was solved by insisting that the Telco turned on CRC4 at their end and then, after an 'init 6', layer two settled down on both systems. I was taught that if you are connecting to a full specification Q.931 circuit, CRC4 should be enabled by default; in the event that one end does not support CRC4 the other end should auto-negotiate back and the circuit should still align without problems. Having said all of this I have yet to see auto-negotiation of CRC4 on any equipment (Public Network or CPE) and suspect that I was not told the truth in the first place... Selection of CRC4 seems to be random from Telco to Telco even on an install by install basis within the same Carrier. It's the first thing to check when new kit appears to be unstable.. HTH Darren -- Comgate Telco>Internetmailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: 07 April 2004 14:59 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Siemens EWSD 13 Hi all, Has anyone got any experience with hooking Asterisk up with a Siemens EWSD 13 switch over a E1/PRI ? We're located in Belgium (Europe) and one of our telecom partners uses this switch. We connected one of our TE410P ports with their switch, but the status light on the TE410P card keeps blinking red. On their side they are getting a DSA (distance service alarm) error, so this normally means the devices 'see' eachother.. but there are still problems with the signalling. Our config below is the same as we are using for MCI, one of our other telecom partners. We tried changing the LBO and timing, but no luck. As you see the signalling is carried over channel 16 (default). TX and RX have also been regularly switched, so no luck.. Their switch is providing the timing. The telecom operator has double checked the asterisk config several times, and it's conform to their setup. The only thing they couldn't find in the Asterisk config is a 'multiframing' option. But I presume this is automatically detected or set by default ? They also tried normal/single(?) framing, but no difference. The card has also been tested with our MCI E1, and works flawlessly, so no hardware issue. Anyone got any further ideas ? Any info or help greatly appreciated! Our config, *** zaptel.conf *** span=1,1,6,ccs,hdb3,crc4,yellow bchan=1-15 bchan=17-31 dchan=16 *** zapata.conf *** [channels] switchtype=euroisdn signalling=pri_cpe pridialplan=unknown group=1 channel => 1-15,17-31 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice Mail Email problem
Ok its probabally something really eaisy im missing. I've searched the archives and voip-info. Asterisk is trying to send the email notification for voice mail. But the log says Invalid sender. Sender = [EMAIL PROTECTED] and not [EMAIL PROTECTED] as assigned in conf file. VM Config: [general] format=gsm|wav49|wav [EMAIL PROTECTED]\n\nYou have received a ${VM_DUR} long message from ${VM_CALLERID}. The Message was left on ${VM_DATE} [bell] 100 => 1234,User1,[EMAIL PROTECTED] <-- Actual file has a valid email. Thanks in advance. Kyle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] res_motv: Request for Comment
YES PLEASE.
Wonderful Stuff! In my opinion just what the project needs. I deployed and
supported many GPL and commercial SmoothWall (firewall) installs and was
forced to poll a web page from time to time to see if any of my customers
needed an urgent security patch applying...not a satisfactory way to manage
many machines deployed across several countries.
The usual caveats about reviewing the 'phone home source code apply of
course as does an opt out for certain Carriers/official organisations that
prefer to remain anonymous.
Regards
Darren
--
Comgate
Telco>Internetmailto:[EMAIL PROTECTED] Behalf Of Mark Spencer
Sent: 07 April 2004 04:31
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] res_motv: Request for Comment
I've been considering the nature of Asterisk, its security, the bug
tracker, and more... And i've come up with an interesting idea: A
"message of the version". The idea is that Asterisk has a compile time
32-bit unsigned int version which is incremented whenever some major new
bug is fixed. When Asterisk starts up (and periodically, maybe once per
day), it sends a packet with the version number to a server at Digium,
along with a message level (INFO,MINOR,MAJOR,CRITICAL) and the Digium
server replies (if it receives the packet, if not, it might get sent again
in a day) with any INFO, MINOR, MAJOR, or CRITICAL messages which are
associated with that version of the code. In this way, an asterisk
administrator could easily see if there were any major issues, critical
security updates, etc, that his system might need to be updated for.
Now, of course, any time you put a "call home" feature in, there are
people who will be concerned about privacy. Clearly it will be able to be
disabled, but I want to run my idea about deployment by everyone here and
see if you guys had some ideas. The idea would be that *new* installs
("make samples") would have the feature turned on for MAJOR level by
default, and that any existing install (e.g. /etc/asterisk/sip.conf
exists, but not /etc/asterisk/motv.conf) would have the file created at
the next "make install" based upon prompting the installer.
Any feedback on:
a) The idea itself -- is it a good one or is it stupid?
b) The way to make it deployed without sneaking a "call home" in on
anybody that doesn't want it?
Thanks!
Mark
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[Asterisk-Users] Presence
Title: [Asterisk-Users] Presence I have to agree. A large number of people are looking for this feature. I have written a web script that can show Agent logged into the system. I think integration/gateway between Asterisk and Jabber would be a amazingly wonderful product. There is always MSN. Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Netural Bay Sydney
Re: [Asterisk-Users] Presence
Shad Mortazavi wrote: I think integration/gateway between Asterisk and Jabber would be a amazingly wonderful product. firefly, while not 100% bug free I think it has this feature, although I haven't played with it enough to work out how to show someone as being online... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk call manager
You should be trying to telnet to the ip address of your asterisk server. Christian Hoffmeyer YottaDot Solutions Huntsville, AL (w) 256.859.4508 (c)256.655.0321 (iax) 700.859.4508 Ask me about Asterisk - Original Message - From: "Jain, Sonal" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, April 07, 2004 2:27 PM Subject: [Asterisk-Users] Asterisk call manager I am trying to setup the call manager and I configured the manager.conf file. When I try to telnet 0.0.0.0 5038 It says trying 0.0.0.0 > > Connected to localhost > Escape character is '^]'. > Asterisk Call Manager/1.0 > Then I type > Action:Login (enter) > Username:sam > Secret:sam > Then I enter twice > > I get Response: error > Message: missing action in request > > I am not sure what it means. > Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WAMi - Windows Asterisk Manager
- Original Message - From: "John Vogel" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, April 07, 2004 5:37 PM Subject: RE: [Asterisk-Users] WAMi - Windows Asterisk Manager > > Doesn't work for me. Connects to Asterisk but says "All extensions are busy > right now" when I try to do anything. Here's what an extension looks like. > Any suggestions? Thanks! > > > Ext 2003 > 2003 > SIP/2003 > from-sip > Assuming you have an extension 2003 in context [from-sip] try: Sip/2003 Christian Hoffmeyer YottaDot Solutions Huntsville, AL (w) 256.859.4508 (c)256.655.0321 (iax) 700.859.4508 Ask me about Asterisk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Presence
They modified iax to include the presence packet but only works on their customized firefly network. I was thinking along the lines of a software app for those of us who use hardware phones but still want to keep TXT chat and presence and perhaps integrated into 1 of the iax soft phones as well to provide a full solution. On Wed, 2004-04-07 at 20:40, Duane wrote: > Shad Mortazavi wrote: > > I think integration/gateway between Asterisk and Jabber would be a > > amazingly wonderful product. > > firefly, while not 100% bug free I think it has this feature, although I > haven't played with it enough to work out how to show someone as being > online... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MySQL CDR
Hi, I'm trying to get CDR recording via MySQL working. I've setup my database and compiled the asterisk-addons from cvs and setup the config file, but when i start asterisk i get this: [cdr_addon_mysql.so] => (MySQL CDR Backend) == Parsing '/etc/asterisk/cdr_mysql.conf': Found Apr 8 11:03:59 WARNING[16384]: config.c:551 cfg_process: parse error: No category context for line 15 of cdr_mysql.conf Apr 8 11:03:59 WARNING[16384]: cdr_addon_mysql.c:267 my_load_module: Unable to load config for mysql CDR's: cdr_mysql.conf Does anyone know what this means? -- jeremy bogan[ [EMAIL PROTECTED] ] segment publishing - design.develop.host ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Presence
William Suffill wrote: They modified iax to include the presence packet but only works on their customized firefly network. I was thinking along the lines of a software app for those of us who use hardware phones but still want to keep TXT chat and presence and perhaps integrated into 1 of the iax soft phones as well to provide a full solution. Question is then, how well does their system work? Already have an IAX2 compatible soft phone with that stuff in it, why not make use of the fact and just work out what needs to be sent to their client... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MySQL CDR
Sounds like an error in your config file. Want to paste the contents in? Thanks... Sean -Original Message- From: Jeremy Bogan [mailto:[EMAIL PROTECTED] Sent: Wednesday, April 07, 2004 8:06 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] MySQL CDR Hi, I'm trying to get CDR recording via MySQL working. I've setup my database and compiled the asterisk-addons from cvs and setup the config file, but when i start asterisk i get this: [cdr_addon_mysql.so] => (MySQL CDR Backend) == Parsing '/etc/asterisk/cdr_mysql.conf': Found Apr 8 11:03:59 WARNING[16384]: config.c:551 cfg_process: parse error: No category context for line 15 of cdr_mysql.conf Apr 8 11:03:59 WARNING[16384]: cdr_addon_mysql.c:267 my_load_module: Unable to load config for mysql CDR's: cdr_mysql.conf Does anyone know what this means? -- jeremy bogan[ [EMAIL PROTECTED] ] segment publishing - design.develop.host ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL CDR
Sounds like an error in your config file. Want to paste the contents in? Thanks... Sorry: ;[global] ;hostname=localhost dbname=asterisk password= user=asterisk ;port=3306 sock=/tmp/mysql.sock userfield=1 -- jeremy bogan[ [EMAIL PROTECTED] ] segment publishing - design.develop.host ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL CDR
;[global] Never mind i'm an idiot. I've commented out [global], whoops! -- jeremy bogan[ [EMAIL PROTECTED] ] segment publishing - design.develop.host ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL CDR
Remove the semi-colon in front of "[global]" HTH, Ryan Thrash On Apr 7, 2004, at 8:15 PM, Jeremy Bogan wrote: Sounds like an error in your config file. Want to paste the contents in? Thanks... Sorry: ;[global] ;hostname=localhost dbname=asterisk password= user=asterisk ;port=3306 sock=/tmp/mysql.sock userfield=1 -- jeremy bogan[ [EMAIL PROTECTED] ] segment publishing - design.develop.host ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Presence
At 8:29 PM -0400 on 4/7/04, Shad Mortazavi wrote:
I have to agree.
A large number of people are looking for this feature. I have
written a web script that can show Agent logged into the system.
I think integration/gateway between Asterisk and Jabber would be a
amazingly wonderful product.
There is always MSN.
Shad Mortazavi
---
Nexus Technical Manager
n|m Nexus Management Inc
Netural Bay
Sydney
My idea for AIM/Jabber/Yahoo integration is below.
Comments and/or programmers are welcome to have at it, and to expand
on my ideas. I have mentioned this to several programmers who
expressed an interest, but I'm sure that lack of time and funding has
kept them from starting on the project, if it indeed is worthwhile.
This is a kludge to some degree, but it uses _already existing_
presence tools to extend Asterisk's functionality, without needing to
modify any client software or hardware.
This is really a one-way presence idea at the moment. There are the
glimmerings of two-way presence (see the "activewhen" keyword) but
this is mostly for CTI outbound notices from an * server to humans
upon some events defined by the administrator. I would see this most
typically used either as a screenpop on an inbound or outbound call,
or perhaps as a voicemail notification tool if the administrator is
clever enough to embed a URL into the string for the instant message
text.
Phase 1: Create a set of programs for Asterisk which allows status
checking of a particular username on a particular instant messaging
system (availability, idle time) and also allows for transmission of
instant messages from Asterisk to other users on those instant
messaging systems (one-way.) The first systems that come to mind
would be AOL's AIM and Yahoo.
Phase 2: Add additional instant message systems: maybe Jabber, MSN.
Allow examination of user's header line (in AOL, at least) and pass
that through the app_imstatus return codes. This would allow me to
specify "mobile:" as the first digits of my status, thus a GotoIf
would be able to know that it should send calls to my cell phone. Or
when I get to work, and shift between my home account ("home: hello,
I'm home") to work ("work: at my desk") then the system will
automatically forward calls appropriately. This might be easy enough
to do in Phase 1, but I'm uncertain.
Future paths:
A true "presence" application for telephony in a large scale method
is lacking today. It may be the case that this could be done by
creating a custom telephony presence presentation application that is
based on an existing (or multiple existing) chat protocols. As an
example, it is possible that I might be able to make my status
message on AIM change from "avail/sip:[EMAIL PROTECTED]" to
"busy/sip:[EMAIL PROTECTED]" every time I pick up the phone;
that could be done programmatically by Asterisk. Then, my friends
who have the custom telephony presence application would see the
little icon beside "pinkycaruthers" go from green to flashing orange.
As soon as I went back to non-busy, they could just click on my icon,
and two things would happen: a password-protected message would get
fired off to THEIR phone system and extension from the presence
application on their desktop, which in turn would be received by an
asterisk-aware application on their Asterisk server, which in turn
would create a spool call to MY phone system from the SIP URI that I
included in my Status message. Presto! We have minimalist call
routing, presence, and click-to-dial - we're just missing the little
app to do it on . The core
message transport protocols all exist; it's just a matter of layers
on top of them. Using standard telephony URI's, we could not just do
this with SIP, but with tel, h323, iax2, anything - it's not limited
to VoIP.
; im.conf
;
; Use of this file implies that you have an active account with one or more
; instant messaging services, and that you probably use an account that is
; dedicated to your Asterisk server so it "knows" what's going on. You may
; need to ensure that any other user id's that you expect to receive messages
; are filtered in such a way that the messages from your Asterisk-specific
; account are permitted through.
;
; username= username of the user on this particular messaging system.
; secret=password for the username
; type= type of connection this is. Each messaging system uses
it's own protocols,
; so we need to specify which one of the protocols we're
using for this particular
; "channel". Current choices are:
; aim - the AOL OSCAR protocol
; yahoo - the Yahoo protocol
; statusmessage= Sets the status message for the user on the chat
server. Visible to other users.
; activewhen= perform a login only when this channel type is valid or
logged in. This is
; reduce unnecessary heartbeat traf
Re: [Asterisk-Users] Spring VON Wrap Up
On 05-04 14:35, Steven Sokol wrote: > TCP/TLS would be used for the SIP messaging which handles call setup, > teardown, and other non-Realtime functions. The voice stream will still be > handled via RTP which is a UDP-based protocol. > > The reason for doing the call setup as TCP is to allow for TLS encryption. > The SIP messages themselves are simply bits of ASCII text (much like SMTP > messages). Currently Asterisk does SIP over UDP only (I think...). In > order to support SIPS (Secure SIP, like HTTPS) we need to build a version of > chan_sip (or chan_sip2 ;-) that supports SIP over TCP. The voice stream > will remain UDP an therefore not succumb to enormous delay. There are some more reasons -- transport of big SIP messages and avoiding network congestion among them. SIP message can get pretty big when XML encoded documents (presence documents, for example) are attached. TCP does not fit everywhere. It is still advantageous to let SIP phones use UDP when communicating with a proxy because the proxy does not have to keep a list of opened connections which is very resource consuming (just consider that you have 10 users using the same proxy -- that can be easily achieved using single server). On the other hand, TCP is useful for proxy-to-proxy communication, especially when there is bigger amount of traffic between proxies. In this case TCP head blocking is really not a problem because the sender gets constant feedback from the remote party and can retransmit the lost segment in a short time. (There was a technical report on this published by Henning Schulzrinne). Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Presence
Duane wrote: William Suffill wrote: They modified iax to include the presence packet but only works on their customized firefly network. I was thinking along the lines of a software app for those of us who use hardware phones but still want to keep TXT chat and presence and perhaps integrated into 1 of the iax soft phones as well to provide a full solution. Question is then, how well does their system work? Already have an IAX2 compatible soft phone with that stuff in it, why not make use of the fact and just work out what needs to be sent to their client... The protocol is quite simple, it's all text messages. S for subscribe to a user's events, T for send a text message I was half way through discussing this with Mark and more specifically adding it to IAX (along with some other cool stuff). Unforunately, I was told to do another project asap but that'll be released next week (stay tuned). My main concern with IAX were you don't know when someone goes offline until their reg expires - no acceptable in presence. Our solution was to keep the registration session open. Keeping the registration session open actually helps everything else fall in place, you can just send messages over that session without requiring setting up a channel, auth and tear down for each message. -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quick Caller ID and Voicemail ?s
[apologies for top-posting] I am very interested in what providers typically take CNAM via ISDN. I have some experience with PRI providers, but I've never heard of one offering that service. If you are a PRI provider in the lower 48 who takes CNAM and can pass that off to the PSTN, please get in touch with me off-list, especially if you handle SIP outbound to your gateways. I may not buy service from you (or maybe I will!) but I'd be very interested in hearing what the particulars are about your offering so I can bring those elements to the table with my current providers. This would be a big benefit for those of us trying to make The Phone System a little more dynamic than the sluggish monster that it currently is. An SQL call and instant re-write for a name is a lot faster/better/more useful than sending a spreadsheet to the carrier every month. JT At 1:34 PM -0400 on 4/7/04, Kyle Thomas wrote: I cannot help you with the .conf files on the * as I am brand new to the * and in the process of compiling the software now. I do know this.. You have to make sure the the generic name IE (information element ) is being populated in the outbound ISDN setup message to Allegiance. If you have a protocol analyzer you could check this , or maybe Allegiance an check this for you. At any rate, they should get this parameter inbound from you ISDN and then pass this parameter to the PSTN (most likely via ISUP). I would get with Allegiance and make sure they are setup to pass CNAM to the PSTN with you. Or you can tell them what ANI/CPN you are sending and they probably store with an SS7 provider , and they can update the SCP for the desired name that you want for any ANI. There might be a cost for this... This last option is the cleanest way to do it. I currently do this for customer's that have PRI PBX's and sit on my switch with a T-1 Kyle On Wed, 2004-04-07 at 12:30, Ryan Thrash wrote: Wow... talk about a detailed response; thanks! In our situation, we've got a T-1 voice PRI from Allegiance Telcom. For the benefit of those of us who aren't as in the know as you are (and who have no affiliation with a CLEC), is there a way to be able to control what gets sent out as our name portion of the Caller ID (even if it means changing what's recorded at Allegiance)? We somehow manage to do so with the number part. In other words, type real slow and mention specific conf files if possible. This is pretty new stuff for me... Thanks again! -- Ryan On Apr 6, 2004, at 7:59 PM, Kyle Thomas wrote: > > SCP=Service control point (database that houses name to number) > SCP DIP = Query to an SCP via the SS7 network > ISUP = SS7 signaling for call setup and teardown (equivalent of > invite,ringing,ok,bye) > IAM = Initial address message (equal to the SIP invite ) > LNP= Local number portability (uses the SS7 network as a backbone). > This > let's people keep thier phone number and switch service providers. > > There is nothing quick about "quick caller id". The far end Telco will > override the name infomration sent to the PSTN and perform thier dips > regardless, overwriting the info you are trying sending out. We are a > CLEC so, > therefore we store, therefore it works.. > > On Tue, 6 Apr 2004, Andrew Kohlsmith wrote: > >>> The terminating telco is doing an SCP dip to thier local SCP's and >>> the >>> database probably does not have that name mapped to this number. >> >>> First thing to do is make sure the generic name ISUP optional >>> paramter is >>> set in the outgoing IAM / ISDN setup from your GW. >> >>> You could also store with an SS7 provider , if these are ported >>> numbers >>> you are sending out make sure that the CNAM field in the LNP line >>> record >>> is set to the point code alias of the provider you are storing with. >>> The >>> terminating switch will first do an LNP dip to see what CNAM alias to > >>> launch the CNAM dip to. If that is not found , will default to the >>> local >>> SCP thus not finding your record. >> >> Ok, and now for the rest of us... >> >> SCP? SCP dip? ISUP? IAM? LNP? >> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kyle Thomas Director of Engineering Monmouth Telephone 732-704-1000 x 130 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Presence
I'm not familiar with the protocol used in Firefly. If that was known then it would be possible to add the functionality to * so anyone can have the simple presences by dialing extensions in their dial plan or crafted packets at a software level. Jabber is already deployed in my organization so I would lean toward integration to that standard as well. On Wed, 2004-04-07 at 21:05, Duane wrote: > William Suffill wrote: > > They modified iax to include the presence packet but only works on their > > customized firefly network. I was thinking along the lines of a software > > app for those of us who use hardware phones but still want to keep TXT > > chat and presence and perhaps integrated into 1 of the iax soft phones > > as well to provide a full solution. > > Question is then, how well does their system work? Already have an IAX2 > compatible soft phone with that stuff in it, why not make use of the > fact and just work out what needs to be sent to their client... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quick Caller ID and Voicemail ?s
Quoting John Todd <[EMAIL PROTECTED]>: > [apologies for top-posting] > > I am very interested in what providers typically take CNAM via ISDN. > I have some experience with PRI providers, but I've never heard of > one offering that service. > > If you are a PRI provider in the lower 48 who takes CNAM and can pass > that off to the PSTN, please get in touch with me off-list, > especially if you handle SIP outbound to your gateways. I may not > buy service from you (or maybe I will!) but I'd be very interested in > hearing what the particulars are about your offering so I can bring > those elements to the table with my current providers. > I'd be interested in this info as well. Ray. - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cell Phone, *, Portability
Currently the plan is to forward all PSTN calls on our 2 incoming PSTN lines & 2 remote toll free's via IAX2 to staff. 3 different delivery methods 1) Users local to the office where the lines come in with GS/PSTN phones 2) IAX2 to remote location * server then Cisco 7960 on that lan 3) IAX2 to remote softphone That assumes they are in the office though. Recently I've been out more than I've been in my office so a VOIP wasn't an option. In this case calls went to me cell and ran up quite alot of minutes. I was wondering how others handle this. Also what carrier you suggest for 2 business cell phones? -- William Suffill ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN BRI solution for USA
I am looking for a ISDN BRI card (u-INTERFACE) to connect * to a US 5ESS switch (Qwest). According to Qwest they support CNAME delivery on their 5ESS switches. Does * chan_capi support CNAME ? Regards. Alfred. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quick Caller ID and Voicemail ?s
At 9:50 PM -0400 4/7/04, Ray Burkholder wrote: From: Ray Burkholder <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Quick Caller ID and Voicemail ?s Quoting John Todd <[EMAIL PROTECTED]>: > [apologies for top-posting] I am very interested in what providers typically take CNAM via ISDN. ...snip... > hearing what the particulars are about your offering so I can bring those elements to the table with my current providers. I'd be interested in this info as well. Ray. I suppose I should bring this around back to a discussion on Asterisk, since this _is_ the Asterisk list and I notice I didn't say anything on my first post: If I can find vendors who can offer this service via PRI, I will try my hardest to find a customer who will pay to have this functionality for outbound name transmission installed into libpri/asterisk, if it isn't there already (which I don't think it is.) There were prior threads about using the UUI header in PRI signalling to move trivial bits of data around; did anything ever come of that? http://lists.digium.com/pipermail/asterisk-dev/2003-September/001747.html JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voice Mail Email problem
It's probably sending the domain as the domain setup on the * server... Change host to somedomain.com and see if that helps... Thanks, -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan Posted At: Wednesday, April 07, 2004 7:32 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] Voice Mail Email problem Subject: [Asterisk-Users] Voice Mail Email problem Ok its probabally something really eaisy im missing. I've searched the archives and voip-info. Asterisk is trying to send the email notification for voice mail. But the log says Invalid sender. Sender = [EMAIL PROTECTED] and not [EMAIL PROTECTED] as assigned in conf file. VM Config: [general] format=gsm|wav49|wav [EMAIL PROTECTED]\n\nYou have received a ${VM_DUR} long message from ${VM_CALLERID}. The Message was left on ${VM_DATE} [bell] 100 => 1234,User1,[EMAIL PROTECTED] <-- Actual file has a valid email. Thanks in advance. Kyle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ISDN BRI solution for USA
I'm also looking for the same thing: ISDN-BRI U interface. Thanks. -brian > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Alfred R. Nurnberger > Sent: Wednesday, April 07, 2004 9:14 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] ISDN BRI solution for USA > > I am looking for a ISDN BRI card (u-INTERFACE) to connect * > to a US 5ESS switch (Qwest). > > According to Qwest they support CNAME delivery on their 5ESS switches. > Does * chan_capi support CNAME ? > > Regards. > Alfred. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Problems with ADIT 600 - latency, loss, etc
Ralph, My experience with the Adit (going on 3 years), it's a solid platform. I've also had one backed into * using MGCP to CMG card for a bit with great success. The thing you said about the bandwidth usage and call quality sucks for both 300KB and 1.5Meg is the clue to this problem being a layer 2 or layer 1 (physical) problem. I would demand a T1 end-to-end, test set to test set check. If there are bipolar violations, check timing along the path, if CRC errors check copper cross connects along the path. Another common problem with inexperienced communications companies is improperly provisioned transport equipment, i.e. one leg of the ckt set to AMI and another leg set to B8ZS, the path may still work but have errors. Bottom line, escalate the problem, if they want to keep your business, they'll get it fixed. Good luck. JR ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
On Fri, 2 Apr 2004 09:32:34 -0500, Adams, Gavin wrote: >We run at 1600x1200, 96 buttons would be useful. > >> -Original Message- >> From: [EMAIL PROTECTED] [mailto:asterisk-users- >> [EMAIL PROTECTED] On Behalf Of Nicolas Gudino >> Sent: Friday, April 02, 2004 9:26 AM >> To: [EMAIL PROTECTED] >> Subject: Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel >> >> Hi Eric, >> >> - Original Message - >> From: "Eric Wieling" <[EMAIL PROTECTED]> >> Sent: Friday, April 02, 2004 11:17 AM >> Subject: Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel >> >> >> > Being able to have more buttons as well as changing the button size >> > would be useful. >> >> What screen resolutions do you use, how many buttons do you need? >> Gee, dual screens, high res Could be great system for hotel etc . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Adtran 850 questions
I just wanted to ask about using Adtran boxes to support analog lines into an Asterisk box. Right now x101p's are just too sensitive to RF noise inside the PC. Going with an external chassis looks like a good, albeit expensive option. It looks like I can use the Digium T1 card into an Adtran 850. The Adtran 850 would need FXO cards. It would also need the echo cancellation module. Is this correct? What other options are out there with hardware level echo cancellation? ...Jeff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP <--> PSTN gateways
Title: SIP <--> PSTN gateways So what are people using these days for SIP or IAX to PSTN gateways. 1. Do any of the standard companies (Packet8, Broadvox, Vonage, etc.) allow you to use your own SIP device (phone or something like *) instead of the interface hardware they usually provide? 2. What about latency and reliability? 3. Finally, do any of the providers deliver more than one call via SIP? In otherwords, if I'm already on a call and another comes in will they attempt to deliver it? Thanks -brian
Re: [Asterisk-Users] chan_oss.c:461: error: too many arguments to function `ast_queue_frame'
On Wed, 7 Apr 2004, Michael T Farnworth wrote: > It appears that the final argument to all these functions (normally a 0 > or 1) has been dropped, but it hasn't been fixed in chan_oss.c or > chan_alsa.c. > The easy fix is just to drop the final arguments for all these functions > and then to kick off the compile again. I had to do the *opposite* of this to get chan_sccp to compile against a freshly-updated CVS of Asterisk and Zaptel. I had to add the extra parameter to these calls in the chan_sccp code. I took it that the extra parameter was added recently, since the last time I compiled the same chan_sccp tree it worked fine. Cheers, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP <--> PSTN gateways
On Wed, 7 Apr 2004, Brian Cuthie wrote: > So what are people using these days for SIP or IAX to PSTN gateways. I'm setting up my own gateways. I'm getting a Cisco 2621XM with a HDV module. I had high hopes for the Ovislink gateways, but they discard proxied SIP requests, for some unknown reason. > 1. Do any of the standard companies (Packet8, Broadvox, Vonage, etc.) allow > you to use your own SIP device (phone or something like *) instead of the > interface hardware they usually provide? Many of the providers use SIP, so even if they don't explicitly tell you the SIP settings to us, it is possible to hookup something else. I've heard that Packet8 gets annoyed when people use self-supplied devices, but it would be possible to hack the code to emulate the device identifier. I've seen an Vonage example posted. > 2. What about latency and reliability? Well, latency is going to be factor of your network and your provider more than anything. If they are bad, it is going to bad. As far as reliablitity goes, perhaps someone else knows more about that. > 3. Finally, do any of the providers deliver more than one call via SIP? In > otherwords, if I'm already on a call and another comes in will they attempt > to deliver it? Good question. I wonder how many of the providers even have the ability to restrict you to one call! I hope the unlimited providers do, because otherwise people using Asterisk as a SIP UA are going to have a field day. Ultimately, you'll need a business SIP connection. There are some providers out there that can do this for LD. > Thanks > > -brian > Tom ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dreaded Caller*ID failed checksum
Caller*ID used to work as some point, but I can't seem to get it going these days. The card is a x101p. I've tried going up and down the rxgain scale. Can the txgain effect it at all? When I plug in a phone into the line with a splitter it can decode caller id with no problems. Reading through the mailing list archives hasn't given me any move clues. Any ideas? ...Jeff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channelized T1, T100P problems
I've been having some problems getting a channelized T1 working with a T100P card. Perhaps someone can help: I have an Eschelon T1 coming into a Vina Integrator box. This box splits out the T1 into an ethernet plug for bandwidth and a secondary T1 which I plug into the T100P card. I've connected the two with a T1 crossover cable. I get a green light on the t100p card and can make outbound calls. Inbound calls are not working as well. Via the console, I can see them come in and start the "s" extension, but then they hang up. Then the second channel accepts a call, then I get a "all circuits busy" message on the phone (from my cell as I call in to test). I'm testing from my cell phone and never hear a ring or any of the asterisk welcome message (even though it's showing in the console). It's as if the call appears to be answered by the card, but actually never is. Changing the "immediate=yes" to "no" changes behavior somewhat. Then I get a message in the console that there is no extension "6" defined, but if I define it, the problem doesn't go away. I am using a channelized T1, not PRI. So we don't have b and d channels. Just e&m signalling. Thanks for your help. Pat Boyle - System: Eschelon T1, channelized (8 channels and the rest for bandwidth) Dell Poweredge 2.4 Celeron Redhat Fedora Asterisk -0.7.2 Zaptel drivers 0.9.0 Here are the config files and console output: --- cat /etc/zaptel.confspan=1,1,0,esf,b8zse&m=1-8loadzone = usdefaultzone=us cat /etc/asterisk/zapata.conf[channels]context=incomingsignalling=em_wgroup=1immediate=yeschannel => 1-8 extensions.conf ;the menu[incoming]exten => s,1,noop; putting a wait here before would cause hangup issues if the person; hung up right after the phone rang. It seems that the card picks; up (simple switch) but doesn't answer until the answer command.; If you hang up before the answer command is executed, there are hangup issues.exten => s,2,Answerexten => s,3,DigitTimeout(5) ; timout for dtmf entries, 5 is defaultexten => s,4,ResponseTimeout(30); hangup after 30 sec of nothingexten => s,5,Background(hello-recording);welcome... then wait for response or timeout ; press 1 for directory; the argument should be the context specified in voicemail.confexten => 1,1,Directory(default) ; if invalid extention, then play invalid message and goto message againexten => i,1,Playback(pbx-invalid)exten => i,2,Goto(s,5) ; on timout hangupexten => t,1,Playback(goodbye)exten => t,2,Hangup exten => h,1,Hangup -- from the console --- *CLI> -- Starting simple switch on 'Zap/1-1' -- Set Digit Timeout to 5 -- Set Response Timeout to 30 -- Playing 'hello-recording' (language 'en')Urgent handler -- Starting simple switch on 'Zap/2-1' -- Set Digit Timeout to 5 -- Set Response Timeout to 30 -- Playing 'hello-recording' (language 'en')Urgent handler -- Hungup 'Zap/1-1'Urgent handlerUrgent handler -- Hungup 'Zap/2-1'Urgent handlerUrgent handler
[Asterisk-Users] Re: indications.conf settings for spain
Saludos. Encantado de ver a alguien español (o no?), en el tiempo que llevo en la lista todavía no lo había visto, o no me había fijado. :) Googlea lo siguiente: "site:lists.digium.com indications conf spain"Sin las comillas claro :) Do You Yahoo!? Todo lo que quieres saber de Estados Unidos, América Latina y el resto del Mundo. Visíta Yahoo! Noticias.
