[Asterisk-Users] Aggressive Echo Suppression
Title: Aggressive Echo Suppression Hi everyone, I have been experimenting with aggressive echo suppression (as defined in zconfig.h) in order to eliminate the infrequent but still disturbing echo that we get with our cisco 7940 phones - asterisk - PRI configuration. Unfortunetely it seems that with the flag on any given call drops after 3 minutes, guaranteed. This happened with both the older production version cvs 03/18/04 which prior to the changes had an over two month uptime, and cvs as of last night. Also just as a matter of interest, anyway to recompile and deploy zaptel / libpri without restarting asterisk? Matt
[Asterisk-Users] Cisco 7940/60 sip downloads
Title: Cisco 7940/60 sip downloads I know this has been hashed out ad nauseum over time but I am interested in getting the SIP firmware for the 7940/60 and the associated minimal contracts for that software but am just getting the runaround from cisco. The wiki was helpful, and some of the comments here have been useful also but I would rather not deal with a reseller if at all possible. Other suggestions / experiences? Off list is fine, thanks very much. Matt
[Asterisk-Users] Avaya Partner Phones to SIP?
Title: Avaya Partner Phones to SIP? I remember someone posting here some time ago about commercial offerings for taking channel banks of Avaya partner phones and turning them into asterisk compatible (SIP?) devices, but I can't seem to find a reference to the hardware manufacturer or specific experiences. Would anyone care to enlighten me? Off list is fine if this is a repeat, thanks very much. Matt
RE: [Asterisk-Users] Advanced queueing
Title: RE: [Asterisk-Users] Advanced queueing Position and hold time announcements/settings are in queues.conf in the later cvs versions. Matt -Original Message- From: Gavin Hamill [mailto:[EMAIL PROTECTED]] Sent: Monday, April 19, 2004 10:48 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Advanced queueing Hullo :) Please be gentle with me, I don't have a working * install, and am just looking for background information. I'm always impressed by companies who implement a queue like You are now number N in the queue. There are currently M agents answering calls, and your call should be answered in approx. O minutes I've seen on http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+additional that Allison has recorded soundfiles to support this style of queue, but how do I make use of them in Asterisk? Is there a pre-written application to implement this type of queue, or would it need to be an AGI-based affair? Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queue out
You can specify a context with single digit extensions for use in a queue, might be in a later cvs release here is the relevant section from the queues.conf ; A context may be specified, in which if the user types a SINGLE ; digit extension while they are in the queue, they will be taken out ; of the queue and sent to that extension in this context. ; ;context = qoutcon Matt On Apr 19, 2004, at 6:03 PM, Jose Maria Guisasola wrote: Please: There is some form so that a user in the queue leaves her (with a digit) and the system execute another command (for example goto a voice mailbox). My version: Asterisk CVS-04/16/04 Thanks in advance -- Jose Mª Guisasola Consultor Técnico CMSI 2002 S.L. ___ 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
Title: RE: [Asterisk-Users] Lucent Phones 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] Non working 800 numbers
Title: Non working 800 numbers Hey guys, I am having a strange problem with certain 800 numbers not working, specifically American Airlines 800-882-8880 800- 843-3000 800- 237-7976 and UPS 800-742-5877 I can't seem to figure out what is causing them not to pick up. Prior to using asterisk on our outbound PRI lines there was no problem. I tried explicitly setting callerid/ani etc on outbound calls, but so far no dice. Has anyone else had a similiar problem? What was the solution? Thanks, Matt
[Asterisk-Users] Pri Errors, Hanging up Owner
Title: Pri Errors, Hanging up Owner Hey guys, Every so often my pri channels degenerate into a non stop series of Mar 15 06:51:50 WARNING[131081]: chan_zap.c:6263 pri_dchannel: Ring requested on channel 1 already in use on span 1. Hanging up owner. Errors. Anyone else having this problem? I see an old reference to updating your cvs, I am using a fairly updated version, as of say a week ago. Anyone have any experience with this / knows what the problem is? Matt
[Asterisk-Users] ISDN PRI A and B, cry for help.
Title: ISDN PRI A and B, cry for help. Does asterisk support PRI protocol version A? Right now I have it working great with B, but in adding some new telco lines I realize that they are spitting out protocol version A, any way to get asterisk to start talking A? Any help would be really appreciated! Matt
[Asterisk-Users] Professional Text to Speech
Title: Professional Text to Speech Hi everyone, I was wondering about peoples experiences with professional text to speech packages, and their integration with Asterisk? Festival at least in the default install sounds... less than perfect. Any suggestions? Matt
RE: [Asterisk-Users] Lucent Definity CallerID {Scanned}
Title: RE: [Asterisk-Users] Lucent Definity CallerID {Scanned}
This doesn't seem to be it, maybe its the definity release I am using but this seems to be set up properly. There must be a flag elsewhere that doesn't pass internal extentions cid informaiton. Any more suggestions?
Matt
-Original Message-
From: htguy [mailto:[EMAIL PROTECTED]]
Sent: Thursday, February 26, 2004 10:31 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Lucent Definity CallerID {Scanned}
Ok, I Clipped this from the tek-tips forum for definity and thought it
might help you with your definity CID issue.
FYI the url I got it from is
http://www.tek-tips.com/gviewthread.cfm/lev2/9/lev3/89/pid/690/qid/742640
-Art
r3jnp1 (Programmer) Jan 22, 2004
Can you send me that email ([EMAIL PROTECTED]) with the
instructions on how to send your DID number to the far ends caller ID
CaNNaBiS (TechnicalUser) Jan 22, 2004
I will tell you how. Its pretty easy.
OK, for example, my extension is 349.
My DID number is 716-897-7349. (notice the last 3 digits of my
DID number match my extension number)
I want 716-897-7349 to show up on the users CID unit.
So I do:
change isdn pub
I make the entry:
Ext Len: 3 (number of digits in my extension)
Ext Code: 3 (the first digit of my extension)
Trk Grp: 12 (my ISDN trunk group)
CPN Prefix: 7168977 (the part I want added to the beginning of
my extension on the CID unit)
CPN Len: 10 (the total number of digits to be displayed on the
CID)
FYI, CID is Caller-ID unit.
#Definity on Efnet
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[Asterisk-Users] Agent Queuing on multiple machines
Title: Agent Queuing on multiple machines Hi, I was wondering if anyone had any experience with agent queueing on multiple machines, because of redundancy in our solution I'm not sure which machine the agents will queue to since they need to log in over zap channels, and which machine the call will come in on, is there a way to make sure that the agent has multiple appearances or otherwise unify the queue? Matt
RE: [Asterisk-Users] Lucent Definity CallerID {Scanned}
Title: RE: [Asterisk-Users] Lucent Definity CallerID {Scanned}
Yeah this combined with the earlier information did it, I think most of the confusion stemmed from running definity release 6. Now it works though, its too bad it can't be set on a per trunk basis though. Thanks very much for the help,
Matt
-Original Message-
From: htguy [mailto:[EMAIL PROTECTED]]
Sent: Friday, February 27, 2004 11:24 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Lucent Definity CallerID {Scanned}
I did come across a PDF explaining how to set up a cisco 3600 series gateway
with a Definity. Maybe it would help. Here is the link
http://www.cisco.com/application/pdf/en/us/guest/products/ps278/c1237/ccmigration_09186a00800e7631.pdf
-Art
- Original Message -
From: Matthew Branton [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, 2004-February-27 10:52
Subject: RE: [Asterisk-Users] Lucent Definity CallerID {Scanned}
This doesn't seem to be it, maybe its the definity release I am using but
this seems to be set up properly. There must be a flag elsewhere that
doesn't pass internal extentions cid informaiton. Any more suggestions?
Matt
-Original Message-
From: htguy [mailto:[EMAIL PROTECTED]]
Sent: Thursday, February 26, 2004 10:31 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Lucent Definity CallerID {Scanned}
Ok, I Clipped this from the tek-tips forum for definity and thought it
might help you with your definity CID issue.
FYI the url I got it from is
http://www.tek-tips.com/gviewthread.cfm/lev2/9/lev3/89/pid/690/qid/742640
-Art
r3jnp1 (Programmer) Jan 22, 2004
Can you send me that email ([EMAIL PROTECTED]) with the
instructions on how to send your DID number to the far ends caller ID
CaNNaBiS (TechnicalUser) Jan 22, 2004
I will tell you how. Its pretty easy.
OK, for example, my extension is 349.
My DID number is 716-897-7349. (notice the last 3 digits of my
DID number match my extension number)
I want 716-897-7349 to show up on the users CID unit.
So I do:
change isdn pub
I make the entry:
Ext Len: 3 (number of digits in my extension)
Ext Code: 3 (the first digit of my extension)
Trk Grp: 12 (my ISDN trunk group)
CPN Prefix: 7168977 (the part I want added to the beginning of
my extension on the CID unit)
CPN Len: 10 (the total number of digits to be displayed on the
CID)
FYI, CID is Caller-ID unit.
#Definity on Efnet
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Re: [Asterisk-Users] Lucent Definity CallerID {Scanned}
We are connecting all the lines via ISDN-PRI to TN767 boards.
Matt
On Feb 26, 2004, at 9:14 PM, htguy wrote:
How are you connecting to the definity? Through analog/digital trunk
ports,
analog station ports or digital station (BRI) ports using an Eicon
Card?
I only have Partner and Magix Systems to test on, but when I get the
new
boards I ordered I'll try it out on the Magix with a few
configurations. If
I can get the Eicon boards working with * then the Magix should be
able to
pass boat loads of information to * and vis-a-versa. (Based on call
center
products installed this should be the same with Definity)
Hmmm I wonder if anyone has success using an Eicon card with * ?
-Art
- Original Message -
From: Matthew Branton [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, 2004-February-26 19:37
Subject: [Asterisk-Users] Lucent Definity CallerID {Scanned}
Hey Guys,
As part of our legacy integration I am trying to get our lucent
definity to pass caller id information from internal stations to
asterisk, I have no problem getting it from lines passed in from the
telco, but the internal stations/vdns etc just don't do it. Anyone
have
any experience in this area? Your input it appreciated,
Matt
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[Asterisk-Users] Lucent Definity CallerID
Hey Guys, As part of our legacy integration I am trying to get our lucent definity to pass caller id information from internal stations to asterisk, I have no problem getting it from lines passed in from the telco, but the internal stations/vdns etc just don't do it. Anyone have any experience in this area? Your input it appreciated, Matt ___ 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] Call Queue wait times
Title: Call Queue wait times Hi Everyone, Is there any specific way to get the current wait time for a queue? If not what is the best way to implement this feature? I would really like to be able to intelligently estimate wait time. Thanks, Matt
RE: [Asterisk-Users] Lucent and ISDN-PRI
Title: RE: [Asterisk-Users] Lucent and ISDN-PRI That document certainly is informative, thanks. I actually went with a tn464F that I happen to have and from the lucent side I have no problem setting it up as a signaling trunk group. Asterisk starts up, registers 1 D-Channel, and 23 B-Channels, but thats as far as I get. When I try to dial the asterisk via the Feature access code I defined on the definity I don't get any sign of a connection. The definity dials, and then waits until timeout at which point I get a busyback. Similarly, if I try to dial out from the Asterisk I get an all busy. I turned on pri intense debug span 1, to see if there were any obvious errors. When I do a dial I get the following traceback: start incredibly long debug message -- [ [02 [02 01 [02 01 01 [02 01 01 38 [02 01 01 38 ] [02 01 01 38 ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000 EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 028 P/F: 0 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter [ [02 [02 01 [02 01 38 [02 01 38 be [02 01 38 be 08 [02 01 38 be 08 02 [02 01 38 be 08 02 80 [02 01 38 be 08 02 80 f8 [02 01 38 be 08 02 80 f8 5a [02 01 38 be 08 02 80 f8 5a 08 [02 01 38 be 08 02 80 f8 5a 08 02 [02 01 38 be 08 02 80 f8 5a 08 02 81 [02 01 38 be 08 02 80 f8 5a 08 02 81 d1 [02 01 38 be 08 02 80 f8 5a 08 02 81 d1 ] [02 01 38 be 08 02 80 f8 5a 08 02 81 d1 ] Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000 EA: 1 N(S): 028 0: 0 N(R): 095 P: 0 9 bytes of data -- ACKing all packets from 94 to (but not including) 95 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 33016/0x80F8) (Terminator) Message type: RELEASE COMPLETE (90) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Invalid call reference value (81), class = Invalid message (5) ] Sending Receiver Ready (29) [ [02 [02 01 [02 01 01 [02 01 01 3a [02 01 01 3a ] [02 01 01 3a ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000 EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 029 P/F: 0 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter [ [02 [02 01 [02 01 3a [02 01 3a be [02 01 3a be 08 [02 01 3a be 08 02 [02 01 3a be 08 02 80 [02 01 3a be 08 02 80 f8 [02 01 3a be 08 02 80 f8 5a [02 01 3a be 08 02 80 f8 5a 08 [02 01 3a be 08 02 80 f8 5a 08 02 [02 01 3a be 08 02 80 f8 5a 08 02 81 [02 01 3a be 08 02 80 f8 5a 08 02 81 d1 [02 01 3a be 08 02 80 f8 5a 08 02 81 d1 ] [02 01 3a be 08 02 80 f8 5a 08 02 81 d1 ] Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000 EA: 1 N(S): 029 0: 0 N(R): 095 P: 0 9 bytes of data -- ACKing all packets from 94 to (but not including) 95 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 33016/0x80F8) (Terminator) Message type: RELEASE COMPLETE (90) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Invalid call reference value (81), class = Invalid message (5) ] Sending Receiver Ready (30) End incredibly long debug message -- Any suggestions? I feel like I am close.. but no cigar. :) Invalid message (5) anyone? I haven't looked at the libpri code but perhaps there is further explanation in there. I'm using pri_cpe channels 1-23, dchan=24, bchan=1-23. Any help is appreciated... thanks again, Matt -Original Message- From: Ken Godee [mailto:[EMAIL PROTECTED]] Sent: Monday, January 19, 2004 5:53 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Lucent and ISDN-PRI Matthew Branton wrote: Hi Everyone, So I have been further exploring the integration of our asterisk server and our lucent definity g3si system. I took the suggestion of setting up an isdn-pri line added the two way tie trunk and the signalling group, but can't seem to get the PRI signalling working on the asterisk correctly. I've set pri type to network on the lucent, and pri_cpe in zapata on the asterisk, but I am a bit confused as to the zaptel settings in this situation. It seems no matter what signaling mode I choose in zaptel.conf (with the exception of clear) I get an error on asterisk startup complaining about requested PRI vs unknown signalling. Any help would be appreciated in getting this working / ironing out some of my conceptual issues. :) I did get the lucent ot work under an em based tie group but that didn't seem to give me any more functionality than I had managed before. Thanks, Matt Matt, You know I'll be following this thread! Found a good reference for G3 isdn-pri you should
[Asterisk-Users] Lucent and ISDN-PRI
Title: Lucent and ISDN-PRI Hi Everyone, So I have been further exploring the integration of our asterisk server and our lucent definity g3si system. I took the suggestion of setting up an isdn-pri line added the two way tie trunk and the signalling group, but can't seem to get the PRI signalling working on the asterisk correctly. I've set pri type to network on the lucent, and pri_cpe in zapata on the asterisk, but I am a bit confused as to the zaptel settings in this situation. It seems no matter what signaling mode I choose in zaptel.conf (with the exception of clear) I get an error on asterisk startup complaining about requested PRI vs unknown signalling. Any help would be appreciated in getting this working / ironing out some of my conceptual issues. :) I did get the lucent ot work under an em based tie group but that didn't seem to give me any more functionality than I had managed before. Thanks, Matt
Re: [Asterisk-Users] Asterisk Integration with Lucent Definity g3si
Come monday I will see if I can get the PRI line working if we have an extra 767 circuit pack. I promise that if/when we get this working I will definitely write up a detailed explanation of the steps involved. Right now we have a partial setup but a fully integrated box seems within reach... any more specifics would be great. Matt On Jan 16, 2004, at 10:47 PM, Ken Godee wrote: PBXtech wrote: We have our G3R setup on a PRI connection. Your trunk group should be set to tie. If anybody would be willing to share just alittle more info on how to set this up it would be great. I've just started thinking about this also. A brief outline would be great, circuit packs used, ds1 settings, trunk group settings and how are you guys setting up the private network to route calls through to/from * ? There's just not a whole lot of info out there on this and I'd rather cut my left nut off, rather than try to talk to Avaya about this. I noticed a spot on the wiki for just this thing, but no ones contributed. http://www.voip-info.org/wiki-Asterisk+legacy+integration Either it's simpler or more complex than I'm making it * via TE410P ISDN-PRI - G3 ISDN-PRI DS1 TN767E (dependancy circuit packs in place, via existing ISDN-PRI) define G3 DS1 assign (tie) trunk group to the G3 DS1 A.)How should one route calls through the G3 out to extentions defined in *? B.) How should one route calls from the * server to/through the G3's ext.s and outbound lines? Any info would be great, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Integration with Lucent Definity g3si
Title: Asterisk Integration with Lucent Definity g3si Hi everyone, We have been working with Asterisk for a while now and would really like to expand its capabilities by fully integrating it with our Definity g3si and are wondering about other peoples experiences with similar setups. Thus far we have only been able to achieve a partial 1 way integration, but ultimately would love to route inbound voip calls to the asterisk out of the lucent. Obviously the tighter the integration we can get the better we can leverage our existing resources and start migrating aspects of our operations to voip entirely. Comments and suggestions from you definity experts out there are very appreciated. The setup: Asterisk 0.7.0 on a vanilla gentoo box with a configured T100P digium card hooked up directly without csu to a TN464 DS1 interface card. With alot of experimentation it connects (With some timing sync issues) with the following settings on the DS1: Bit rate: 1.544 line coding: ami-basic line compensation: 1 framing mode: d4 signalling mode: robbed bit Interface companding: mulaw idle code: slip detect: n near-end CSU type: other We then set up a trunk group, but I believe because of the limitations of the card/setup we can only use group type: co which limits us to outbound on the trunk, or inbound to 1 mapped extension, no DID or more advanced switching. The temporary solution to this has been to use some of the lucents built in vectoring capabilities to send specific extensions to the asterisk. Any recommendations for increasing our integration? Cards/setup etc? Thanks very much. Being able to send voip calls on the asterisk to queues on the Lucent would be fantastic. Matt
