Re: [asterisk-users] MOH question w/Cisco 79xx phones
I think in your SIPDefault.cnf you disable VAD enable_vad: 0 ; VAD setting 0-disable (Default), 1-enable Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary Sent: Friday, June 29, 2007 9:28 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] MOH question w/Cisco 79xx phones Hi Everyone Got a newbie type question regarding MOH Cisco phones. I'm still new to Asterisk (very new in fact) built up a AsteriskNOW box just to get something going. My simple test system has just 3 Cisco phones a 7905, 7940 7960. - Everything's running SIP. The 3 phones can call each other fine. - Can even leave (and retreive) voicemail messages. - No problems. My problem is with MOH (Music On Hold). - The 7940/7960 are somewhat okay, but the 7905 is another story. When a is call from a 7940/7960 is placed on 'hold' (by the calling party), MOH starts up on the 7940/7960, plays for about a second or two, then drops out for about a second or so, then continues. - After that, it continues to play okay. But when a call from the 7905 is placed on 'hold' (by the calling party), MOH starts up on the 7905, plays for a second or two, drops out for a sec, starts again for a sec or so, drops out, starts back up, drops out, etc., etc., etc Just up and down. - Kinda' like a Yo-Yo. Also - When the call from the 7905 is placed on hold, I see the following warning at the Asterisk CLI: [Jun 29 22:18:28] NOTICE[3376]: rtp.c783 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 10.0.0.110 I don't see this warning when the 7940/7960 is playing MOH. I'm using basic default settings for just about everything. - Could this be with the RTP config? - The 7905 Audio settings? Anybody have a clue? Thanks in advance. Gary Guthary ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] forwarding loop not detected
Asterisk 1.2.16 I have an extension 102 with a Polycom 430 I am trying to protect against forwarding loops If I set the phone to forward the line to itself, extension 102 I get the following -- Got SIP response 302 Moved Temporarily back from 206.83.240.18 -- Now forwarding Local/[EMAIL PROTECTED],2 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/exten-mycontext-102-094c2c08) -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/exten-mycontext-102) in new stack -- Called exten-mycontext-102 -- Got SIP response 302 Moved Temporarily back from 206.83.240.18 -- Now forwarding Local/[EMAIL PROTECTED],2 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/exten-mycontext-102-095bfef8) -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/exten-mycontext-102) in new stack -- Called exten-mycontext-102 -- Got SIP response 302 Moved Temporarily back from 206.83.240.18 -- Now forwarding Local/[EMAIL PROTECTED],2 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/exten-mycontext-102-09495990) -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/exten-mycontext-102) in new stack -- Called exten-mycontext-102 Looping for a long time then the next entry in the dial plan kicks in (Voicemail) after a ton of those Dialplan: exten = 102,1,Dial(SIP/exten-mycontext-102) exten = 102,n,Voicemail([EMAIL PROTECTED]) Forwarding to other extensions and outside numbers works fine, just not to itself. How can I protect against such loops? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PROGRESS code
I have a PRI switch type national
Asterisk 1.2.16
Zaptel 1.2.15
If I call an invalid number I get
* PROGRESS with cause code 28 received
Asterisk continues to attempt to connect the call until the timeout is
reached and I hear ringing.
I want to capture the progress code, which I thought was in HANGUPCAUSE
but when I NoOp that variable it's always 16 when I dial an invalid
number...not 28
Also, I don't see how to immediately indicate the number is invalid,
without waiting for the channel to automatically hang up.
Is that just the way it works...I gotta wait?
Why is HANGUPCAUSE 16 but I get Progress cause code 28? 28 is clearly
correct because 11 is an invalid number format.
exten = _.,1,Dial(Zap/g1/${EXTEN})
exten = _.,n,NoOp(Dial Status is ${DIALSTATUS})
exten = _.,n,NoOp(Hang Up Clause is ${HANGUPCAUSE})
exten = _.,n,Congestion
-- Executing Dial(SIP/x.x.x.x-090bec30, Zap/g1/11) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/11
-- Zap/1-1 is proceeding passing it to SIP/x.x.x.x-090bec30
-- PROGRESS with cause code 28 received
-- Zap/1-1 is making progress passing it to SIP/x.x.x.x-090bec30
-- Channel 0/1, span 1 got hangup request
-- Hungup 'Zap/1-1'
== No one is available to answer at this time (1:0/0/0)
-- Executing NoOp(SIP/x.x.x.x-090bec30, Dial Status is NOANSWER)
in new stack
-- Executing NoOp(SIP/x.x.x.x-090bec30, Hang Up Cause is 16) in
new stack
-- Executing Congestion(SIP/x.x.x.x-090bec30, ) in new stack
== Spawn extension (pri-only, 11, 4) exited non-zero on
'SIP/x.x.x.x-090bec30'
-- Executing Hangup(SIP/x.x.x.x-090bec30, ) in new stack
Bill
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom call parking feature and Asterisk callparking
Using the Park button actually requires more work than just doing an attended
transfer to the park extension
Anyway, use the ParkAndAnnounceFunction, here's an example
exten =
callpark,n,ParkAndAnnounce(pbx-transfer:PARKED|10|SIP/${DIALEDPEERNUMBER}|app-directfrompark,*81${DIALEDPEERNUMBER},1)
-Original Message-
From: [EMAIL PROTECTED] on behalf of Stephen Bosch
Sent: Fri 3/9/2007 4:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Polycom call parking feature and Asterisk callparking
Hi:
I want to make parking calls easier for my hard-working users. Is there
a way to make the Polycom call park feature work with Asterisk?
In case it just works out of the box, I haven't tried it yet; but the
call park feature isn't enabled on the Polycom phones by default.
-Stephen-
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom call parking feature and Asteriskcallparking
When you use the Park button, on some phones you have to hit More to
get it.
Then when you park it, it calls back and tells you the extension...so
you have to hang up then pick up again.
Callpark apparently is a valid extension!
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen
Bosch
Sent: Saturday, March 10, 2007 12:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom call parking feature and
Asteriskcallparking
Bill Gibbs wrote:
Using the Park button actually requires more work than just doing an
attended transfer to the park extension
Does it? How does it work exactly?
What I would expect:
Press the Park button; hear the announcement; hang up
The way it works now:
Press the Transfer button; dial 700; press Send; hear the announcement;
press Transfer; hang up
Anyway, use the ParkAndAnnounceFunction, here's an example
exten =
callpark,n,ParkAndAnnounce(pbx-transfer:PARKED|10|SIP/${DIALEDPEERNUMBER
}|app-directfrompark,*81${DIALEDPEERNUMBER},1)
Is callpark a valid extension?
Thanks,
-Stephen-
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Sender phone ringing while recipient talking
Are you using the option r in your Dial string? If so, remove it. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nathan Bell Sent: Thursday, March 08, 2007 1:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sender phone ringing while recipient talking I've had asterisk running for about a month now between our PBX and our T1, and everything seems fine but for one simple nit-pick: When a call to the outside workd is made, and if the recipient picks up while a the sender's phone is still relaying the ring, the sender won't be heard until after the ring stops. This often translates a simple hello? into a lo? or even *long pause* hello, is anyone there? Is there a way to immediately stop the ring when a pickup is detected? Thanks, Nathan Bell ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip show channels
Behavior on Asterisk 1.2.12, 1.2.15, 1.2.16 sip show channels Always tends to show 100+ lines such as 192.168.1.241(None) 2e2872da-1d 00101/21507 unkn No Rx: REGISTER Never seem to go away 198 total peers on this server All devices are behind NAT Registration expirations between 30secs to 2 minutes to help keep NAT open Should I extend the registration expiration to maybe 5 minutes? Will that keep the NAT holes on the remote router open long enough to receive calls? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 1.4 - SLA
I think it has something to do with hints...I can't seem to subscribe to anything now with 1.4 vs 1.2, even with a normal non SLA setup. My phone/config that works with 1.2, so I know hints work with the phone and firmware and with NAT at least on 1.2. I did a fresh 1.4 install (and I did a make samples so I had something to work off of) sip show subscriptions shows 0 active show hints: [EMAIL PROTECTED] : SIP/2404366402State:Idle Watchers 0 If I run the default demo app, show hints still shows Idle. My Buddies key in the Polycom, which is watching the proper sip hint (works in 1.2) shows the extension to be Offline. Sip.conf [general] allowsubscribe=yes subscribecontext=default notifyringing=yes notifyhold=yes limitonpeers=yes (I tried with and without the above values commented out, as well as specifically in my device peer definition) [2404366402] type=friend secret=blahededah nat=yes host=dynamic canreinvite=no context=default qualify=yes extensions.conf [default] exten = 2404366402,hint,SIP/2404366402 ...etc... My mac-directory.xml ..snip... item lnmyself/ln fn/fn ct2404366402/ct sd/sd rt/rt dc/ ad0/ad ar0/ar bw1/bw bb0/bb /item ...snip... I also tried in the ct[EMAIL PROTECTED]/ct Let's pretend 1.1.1.1 is my firewall that the Polycom is behind 2.2.2.2 is my 1.4.1 test Asterisk server --- SIP read from 1.1.1.1:60671 --- SUBSCRIBE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116;branch=z9hG4bK256e271aEC1CEA7B From: Line 1 sip:[EMAIL PROTECTED];tag=447AE7-653FB66C To: sip:[EMAIL PROTECTED] CSeq: 1 SUBSCRIBE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094 Max-Forwards: 70 Expires: 3600 Content-Length: 0 - --- (13 headers 0 lines) --- Creating new subscription Sending to 192.168.1.116 : 5060 (no NAT) Found peer '2404366402' --- Transmitting (NAT) to 1.1.1.1:60671 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.116;branch=z9hG4bK256e271aEC1CEA7B;received=1.1.1.1 From: Line 1 sip:[EMAIL PROTECTED];tag=447AE7-653FB66C To: sip:[EMAIL PROTECTED];tag=as3123a96d Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=7a544b2b Content-Length: 0 --- SIP read from 1.1.1.1:60671 --- SUBSCRIBE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116;branch=z9hG4bK3f8d777548EC8ED2 From: Line 1 sip:[EMAIL PROTECTED];tag=447AE7-653FB66C To: sip:[EMAIL PROTECTED] CSeq: 2 SUBSCRIBE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094 Authorization: Digest username=2404366402, realm=asterisk, nonce=7a544b2b, uri=sip:[EMAIL PROTECTED]:5060, response=404b224f5abbdc3793d4df45ee2ffa59, algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 - --- (14 headers 0 lines) --- Creating new subscription Sending to 1.1.1.1 : 60671 (NAT) Found peer '2404366402' Looking for 2404366402 in default (domain 2.2.2.2) --- Transmitting (NAT) to 1.1.1.1:60671 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.116;branch=z9hG4bK3f8d777548EC8ED2;received=1.1.1.1 From: Line 1 sip:[EMAIL PROTECTED];tag=447AE7-653FB66C To: sip:[EMAIL PROTECTED];tag=as3123a96d Call-ID: [EMAIL PROTECTED] CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russell Bryant Sent: Tuesday, March 06, 2007 1:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.4 - SLA Bill Gibbs wrote: I have been using 2 Polycom 430s so far. I can get incoming calls just fine (both phones ring on line 1). However it doesn't appear to seize the line, so if a call is on the one phone, I can still pick up line 1 on the other and dial - and it's reflected in the connected call. I assume that's related to the hint/subscription issue Lacy indicated as well. sip show subscriptions shows nothing. If you see no subscriptions, then the phones will not dispaly the state of the line at all. In regards to still allowing you to dial when all lines are busy, do you have your phones set up to automatically dial when you go off-hook? In this SLA setup, you should not allow any
RE: [asterisk-users] 1.4 - SLA
I assume my SUBSCRIBE issue for hints has something to do with this bug http://bugs.digium.com/view.php?id=9168 Bill snipped previous emails for readability ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 1.4 - SLA
I have been using 2 Polycom 430s so far. I can get incoming calls just fine (both phones ring on line 1). However it doesn't appear to seize the line, so if a call is on the one phone, I can still pick up line 1 on the other and dial - and it's reflected in the connected call. I assume that's related to the hint/subscription issue Lacy indicated as well. sip show subscriptions shows nothing. I just started playing with it this morning however...still playing around w/ the configs. One odd thing, I keep seeing some weirdness: [Mar 5 13:10:52] VERBOSE[9381] logger.c: Looking for 105 in default (domain x.x.x.x) And also Looking for 103 Yet I have no idea where those values are coming from! I am running 1.6.7. Here is a snippet of the phone config from one of the phones: reg reg.1.displayName=Line 1 reg.1.address=station2_line1 reg.1.label=Line 1 reg.1.type=shared reg.1.thirdPartyName=2404366402-2 reg.1.auth.userId=2404366402-2 reg.1.auth.password=1234 reg.1.server.1.address= reg.1.server.1.port= reg.1.server.1.transport=DNSnaptr reg.1.server.2.transport=DNSnaptr reg.1.server.1.expires=60 reg.1.server.1.register=1 I noticed that I had to set the reg.x.address field to the stationX_lineX value or the phone wouldn't fill in the icon image...but it would accept cals. Still not completely clear but I am making progress! Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russell Bryant Sent: Friday, March 02, 2007 6:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.4 - SLA Lacy Moore - Aspendora wrote: Russell, I don't have any specifics at this time. I need to dig a little further. I'm thinking the autocontext is what is giving me fits. I can receive calls and place calls, but the hint status is not working. It currently registers as a hint showing not in use. It does not show in use. If you aren't seeing any lights change on the phones when calls are going on, check sip show subscriptions at the CLI. If the phones have not properly subscribed to the right extensions, you won't see anything. I ended up using some of the config from the bottom of the sla.txt file. The sample file may be missing the template section. The sample config does not match the config in the sla.txt. I couldn't get the sample config to work at all. Again, hopefully over the weekend I'll be able to get more information. You are correct. The sample configuration is missing the template. I will add it now. However, I just made the tarballs for 1.4.1, so this config fix didn't make it in. Using the config in the sample file, the hint status was working. I could see the line ringing, but I could not answer the lines or place calls. Using the config from the sla.txt file, I could place calls and receive calls, but the hints never showed any activity, just always not in use. As I noted earlier, check your sip show subscriptions to make sure the phones are subscribed to the right thing. Another helpful thing that you can use for debugging is to look at the output of sla show stations. You can see the state of each line appearance on each station. This should correspond with what you see on the phone ... unless there is a problem, of course. If possible, could you provide the config that you've used for testing? I'm testing using Polycom phones to try to keep things simple. I'm assuming you are using a Polycom. I have been testing with a variety of different phones. I have not tested all of the Polycom models, yet. This is one of the things we're going to have to work through. I would like to document issues with specific phones in sla.txt as we come across them. The configuration I'm using for testing looks just like the stuff in configs/sla.conf.sample. Essentially, it is: [line1] type=trunk device=Zap/3 autocontext=line1 [line2] type=trunk device=Zap/4 autocontext=line2 [station](!) type=station autocontext=sla_stations trunk=line1 trunk=line2 [station1] (station) device=SIP/station1 [station2](station) device=SIP/station2 [station3](station) device=SIP/station3 Thanks for providing some feedback on this. You are the first one to say anything about it. :) I am very eager to get everything working well so that everyone is happy. Just please be patient as I work through the initial flood of reports since it is just now getting out in the field. -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 1.4 - SLA
Here is the debug output of the SUBSCRIBE request I am sure it has something to do with the way I am attempting to setup the Polycom for shared appearances... Nat=yes is set in the peer. I don't get these weird messages when connecting with a private line appearance. --- SIP read from x.x.x.x:60671 --- SUBSCRIBE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116;branch=z9hG4bKf2bac598B416612D From: Line 1 sip:[EMAIL PROTECTED];tag=259528A1-76B251C6 To: sip:[EMAIL PROTECTED] CSeq: 1 SUBSCRIBE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094 Max-Forwards: 70 Expires: 3600 Content-Length: 0 - [Mar 5 14:25:02] VERBOSE[9835] logger.c: --- (13 headers 0 lines) --- [Mar 5 14:25:02] VERBOSE[9835] logger.c: Sending to 192.168.1.116 : 5060 (no NAT) [Mar 5 14:25:02] VERBOSE[9835] logger.c: Found no matching peer or user for 'x.x.x.x:60671' [Mar 5 14:25:02] VERBOSE[9835] logger.c: Looking for 103 in default (domain x.x.x.x) [Mar 5 14:25:02] VERBOSE[9835] logger.c: --- Transmitting (no NAT) to 192.168.1.116:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.116;branch=z9hG4bKf2bac598B416612D;received=x.x.x.x From: Line 1 sip:[EMAIL PROTECTED];tag=259528A1-76B251C6 To: sip:[EMAIL PROTECTED];tag=as4d77da56 Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Monday, March 05, 2007 1:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] 1.4 - SLA I have been using 2 Polycom 430s so far. I can get incoming calls just fine (both phones ring on line 1). However it doesn't appear to seize the line, so if a call is on the one phone, I can still pick up line 1 on the other and dial - and it's reflected in the connected call. I assume that's related to the hint/subscription issue Lacy indicated as well. sip show subscriptions shows nothing. I just started playing with it this morning however...still playing around w/ the configs. One odd thing, I keep seeing some weirdness: [Mar 5 13:10:52] VERBOSE[9381] logger.c: Looking for 105 in default (domain x.x.x.x) And also Looking for 103 Yet I have no idea where those values are coming from! I am running 1.6.7. Here is a snippet of the phone config from one of the phones: reg reg.1.displayName=Line 1 reg.1.address=station2_line1 reg.1.label=Line 1 reg.1.type=shared reg.1.thirdPartyName=2404366402-2 reg.1.auth.userId=2404366402-2 reg.1.auth.password=1234 reg.1.server.1.address= reg.1.server.1.port= reg.1.server.1.transport=DNSnaptr reg.1.server.2.transport=DNSnaptr reg.1.server.1.expires=60 reg.1.server.1.register=1 I noticed that I had to set the reg.x.address field to the stationX_lineX value or the phone wouldn't fill in the icon image...but it would accept cals. Still not completely clear but I am making progress! Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russell Bryant Sent: Friday, March 02, 2007 6:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.4 - SLA Lacy Moore - Aspendora wrote: Russell, I don't have any specifics at this time. I need to dig a little further. I'm thinking the autocontext is what is giving me fits. I can receive calls and place calls, but the hint status is not working. It currently registers as a hint showing not in use. It does not show in use. If you aren't seeing any lights change on the phones when calls are going on, check sip show subscriptions at the CLI. If the phones have not properly subscribed to the right extensions, you won't see anything. I ended up using some of the config from the bottom of the sla.txt file. The sample file may be missing the template section. The sample config does not match the config in the sla.txt. I couldn't get the sample config to work at all. Again, hopefully over the weekend I'll be able to get more information. You are correct. The sample configuration is missing the template. I will add it now. However, I just made the tarballs for 1.4.1, so this config fix didn't make it in. Using the config in the sample file, the hint status was working. I could see the line ringing, but I could not answer the lines or place calls. Using the config from the sla.txt file, I could place calls and receive calls, but the hints never showed any activity, just always not in use. As I noted earlier, check your sip show subscriptions to make sure the phones are subscribed to the right thing. Another helpful thing that you can use for debugging is to look at the output of sla show stations
RE: [asterisk-users] 1.4 - SLA
Sorry to reply to myself, once again onn the list, but since SLA is new I figured I should answer my own question before anyone else gets confused...I completely forgot about my -directory.xml defaults...so that's where all these bogus SUBSCRIBE requests were coming from. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Monday, March 05, 2007 2:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] 1.4 - SLA Here is the debug output of the SUBSCRIBE request I am sure it has something to do with the way I am attempting to setup the Polycom for shared appearances... Nat=yes is set in the peer. I don't get these weird messages when connecting with a private line appearance. --- SIP read from x.x.x.x:60671 --- SUBSCRIBE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116;branch=z9hG4bKf2bac598B416612D From: Line 1 sip:[EMAIL PROTECTED];tag=259528A1-76B251C6 To: sip:[EMAIL PROTECTED] CSeq: 1 SUBSCRIBE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094 Max-Forwards: 70 Expires: 3600 Content-Length: 0 - [Mar 5 14:25:02] VERBOSE[9835] logger.c: --- (13 headers 0 lines) --- [Mar 5 14:25:02] VERBOSE[9835] logger.c: Sending to 192.168.1.116 : 5060 (no NAT) [Mar 5 14:25:02] VERBOSE[9835] logger.c: Found no matching peer or user for 'x.x.x.x:60671' [Mar 5 14:25:02] VERBOSE[9835] logger.c: Looking for 103 in default (domain x.x.x.x) [Mar 5 14:25:02] VERBOSE[9835] logger.c: --- Transmitting (no NAT) to 192.168.1.116:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.116;branch=z9hG4bKf2bac598B416612D;received=x.x.x.x From: Line 1 sip:[EMAIL PROTECTED];tag=259528A1-76B251C6 To: sip:[EMAIL PROTECTED];tag=as4d77da56 Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Monday, March 05, 2007 1:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] 1.4 - SLA I have been using 2 Polycom 430s so far. I can get incoming calls just fine (both phones ring on line 1). However it doesn't appear to seize the line, so if a call is on the one phone, I can still pick up line 1 on the other and dial - and it's reflected in the connected call. I assume that's related to the hint/subscription issue Lacy indicated as well. sip show subscriptions shows nothing. I just started playing with it this morning however...still playing around w/ the configs. One odd thing, I keep seeing some weirdness: [Mar 5 13:10:52] VERBOSE[9381] logger.c: Looking for 105 in default (domain x.x.x.x) And also Looking for 103 Yet I have no idea where those values are coming from! I am running 1.6.7. Here is a snippet of the phone config from one of the phones: reg reg.1.displayName=Line 1 reg.1.address=station2_line1 reg.1.label=Line 1 reg.1.type=shared reg.1.thirdPartyName=2404366402-2 reg.1.auth.userId=2404366402-2 reg.1.auth.password=1234 reg.1.server.1.address= reg.1.server.1.port= reg.1.server.1.transport=DNSnaptr reg.1.server.2.transport=DNSnaptr reg.1.server.1.expires=60 reg.1.server.1.register=1 I noticed that I had to set the reg.x.address field to the stationX_lineX value or the phone wouldn't fill in the icon image...but it would accept cals. Still not completely clear but I am making progress! Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russell Bryant Sent: Friday, March 02, 2007 6:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.4 - SLA Lacy Moore - Aspendora wrote: Russell, I don't have any specifics at this time. I need to dig a little further. I'm thinking the autocontext is what is giving me fits. I can receive calls and place calls, but the hint status is not working. It currently registers as a hint showing not in use. It does not show in use. If you aren't seeing any lights change on the phones when calls are going on, check sip show subscriptions at the CLI. If the phones have not properly subscribed to the right extensions, you won't see anything. I ended up using some of the config from the bottom of the sla.txt file. The sample file may be missing the template section. The sample config does not match the config in the sla.txt. I couldn't get the sample config to work at all. Again, hopefully over the weekend I'll be able to get more information. You are correct. The sample configuration is missing the template. I will add it now. However, I just made the tarballs for 1.4.1, so this config fix didn't make
RE: [asterisk-users] Fax with T.38
I am waiting for the powers that be to get a dual port PRI card at this time. I think a dial-peer will only need to look similar to this on the Cisco: dial-peer voice 10 voip destination-pattern WHATEVER session protocol sipv2 session target ipv4:openpbx ip dtmf-relay sip-notify rtp-nte fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco Since that's basically what you need to do voice, all this adds is the T38 line. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jbebeau Sent: Saturday, February 24, 2007 3:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Fax with T.38 Hi Bill, I'm in exactly the same boat with T.38 and OpenPBX. I too think the Cisco-T.38 Gateway is the most practical at this moment. Where are you on testing this and can you share the 3660 config? In researching the CIsco/voice, there is a TON of hardware options you need, or so it seems. Jon -Original message- From: Bill Gibbs [EMAIL PROTECTED] Date: Thu, 22 Feb 2007 15:02:18 -0500 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: RE: [asterisk-users] Fax with T.38 Ray, I have been playing with OpenPBX. My core servers are Asterisk so I was playing around with their T38Gateway application. Long story short - I can get the ATA (behind NAT) to talk T38 to the rxfax app on an OpenPBX server but the gateway feature of that product is still under development so I was sending IAX calls to it and it would try to talk T38 to my ATA (behind NAT or public IP) and eventually the call would fail. Clearly T38 was working though, debug output was full of T38 talk. However the wiki clearly states it's experimental still. I personally have decided to go with a 2nd PRI port to a 3660 I have on hand that will do T38 SIP. I am going to set that up to talk to * 1.4.0 and do T38 pass through. I to will be doing NAT for the ATAs so...hopefully it will work. We shall see. So my call flow will be PRI - Asterisk 1.2.x Out the 2nd PRI to the 3660 3660 dial-peer, with T38 fax settings talk SIP to Asterisk 1.4.x then t38 pass through to my ATA. I have the 3660 there to take the call via TDM and convert to T38. I only have a single PRI which is why I don't want to have to purchase other lines dedicated to a T38 faxserver, and this will give me the ability to use my DIDs already assigned. That's how I plan to set it up. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ray Jackson Sent: Wednesday, February 21, 2007 10:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax with T.38 Could anybody give me an authoritative answer on whether Asterisk can support T.38 pass-through when the clients are behind NAT? We have Asterisk servicing clients behind NAT (with nat=route, canreinvite=no) and would love to get T.38 going but have had no luck so far. The following case: http://bugs.digium.com/view.php?id=7844 ...suggests that T.38 *does* now work for clients behind NAT but I have the latest SVN trunk but still cannot get it to work? On the other side I have seen on this list only 2 weeks or so ago: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg172556.html This suggests that T.38 does *NOT* work behind NAT? So, can anybody save me the trouble and tell me how it is. Am I on a hiding to nothing trying to get T.38 going with NAT? Please put me out of my misery! :) Cheers, Ray PS. Does anybody know whether OpenPBX would support T.38 and NAT configurations? This was my backup plan if I couldn't get it to go in Asterisk. Thomas Deillon wrote: Yes, the canreinvite means Re invite, but there is a consequence in Asterisk configuration. For sure, all the signalisation traffic will go through the asterisk … but for the RTP traffic? If canreinvite = No, all RTP traffic will go through the Asterisk (useful for NATed phoned without ALG/STUN/…) If canreinvite = Yes, the phones will try to exchange RTP packets directly. Do you thing there is a way to allow Re Invite (because you’re right) without the RTP consequence? Thanks a lot for your help, Thomas *De :* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *De la part de* Rajnish Jain *Envoyé :* lundi, 19. février 2007 16:25 *À :* Asterisk Users Mailing List - Non-Commercial Discussion *Objet :* Re: [asterisk-users] Fax with T.38 A T.38 fax call typically begins as a normal voice media call. The call then dynamically switches over T.38 image media on detection of fax handshake tones. The dynamic modification of session from audio to image
RE: [asterisk-users] Polycom SIP 501 Transfer Question
Yeah but I think the caller ID issue still remains. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Friday, February 23, 2007 11:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom SIP 501 Transfer Question In later 1.6.x firmwares there is a config option for allow transfer on proceeding that basically allows you to do a blind transfer by just hitting the transfer key again rather than having to select Blind. Shawn Kelley wrote: I know this is not a Polycom support forum, but I also know there are a lot of you with a great deal of Polycom experience. Is there anyway to remove the Attended Transfer but keep the Blind transfer? Or better yet, just swap the two soft buttons locations? I know you can remap the Hard buttons, but what about the soft buttons? The reason I need this is my users can't get it through their head that they need to announce the call if they use the normal aka Attended transfer before the press the transfer button again to complete it. I know if they would just use the Blind transfer we would have no problems, but since the Blind transfer is on the second set of screen soft buttons they aren't smart enough to find it I guess. The problem with them using Attended Transfer is CallerID shows up as theirs, when in reality they have already press the transfer button a second time. We then don't answer the phone professionally since we think that it is our employee calling us. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fax with T.38
Ray, I have been playing with OpenPBX. My core servers are Asterisk so I was playing around with their T38Gateway application. Long story short - I can get the ATA (behind NAT) to talk T38 to the rxfax app on an OpenPBX server but the gateway feature of that product is still under development so I was sending IAX calls to it and it would try to talk T38 to my ATA (behind NAT or public IP) and eventually the call would fail. Clearly T38 was working though, debug output was full of T38 talk. However the wiki clearly states it's experimental still. I personally have decided to go with a 2nd PRI port to a 3660 I have on hand that will do T38 SIP. I am going to set that up to talk to * 1.4.0 and do T38 pass through. I to will be doing NAT for the ATAs so...hopefully it will work. We shall see. So my call flow will be PRI - Asterisk 1.2.x Out the 2nd PRI to the 3660 3660 dial-peer, with T38 fax settings talk SIP to Asterisk 1.4.x then t38 pass through to my ATA. I have the 3660 there to take the call via TDM and convert to T38. I only have a single PRI which is why I don't want to have to purchase other lines dedicated to a T38 faxserver, and this will give me the ability to use my DIDs already assigned. That's how I plan to set it up. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ray Jackson Sent: Wednesday, February 21, 2007 10:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax with T.38 Could anybody give me an authoritative answer on whether Asterisk can support T.38 pass-through when the clients are behind NAT? We have Asterisk servicing clients behind NAT (with nat=route, canreinvite=no) and would love to get T.38 going but have had no luck so far. The following case: http://bugs.digium.com/view.php?id=7844 ...suggests that T.38 *does* now work for clients behind NAT but I have the latest SVN trunk but still cannot get it to work? On the other side I have seen on this list only 2 weeks or so ago: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg172556.html This suggests that T.38 does *NOT* work behind NAT? So, can anybody save me the trouble and tell me how it is. Am I on a hiding to nothing trying to get T.38 going with NAT? Please put me out of my misery! :) Cheers, Ray PS. Does anybody know whether OpenPBX would support T.38 and NAT configurations? This was my backup plan if I couldn't get it to go in Asterisk. Thomas Deillon wrote: Yes, the canreinvite means Re invite, but there is a consequence in Asterisk configuration. For sure, all the signalisation traffic will go through the asterisk … but for the RTP traffic? If canreinvite = No, all RTP traffic will go through the Asterisk (useful for NATed phoned without ALG/STUN/…) If canreinvite = Yes, the phones will try to exchange RTP packets directly. Do you thing there is a way to allow Re Invite (because you’re right) without the RTP consequence? Thanks a lot for your help, Thomas *De :* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *De la part de* Rajnish Jain *Envoyé :* lundi, 19. février 2007 16:25 *À :* Asterisk Users Mailing List - Non-Commercial Discussion *Objet :* Re: [asterisk-users] Fax with T.38 A T.38 fax call typically begins as a normal voice media call. The call then dynamically switches over T.38 image media on detection of fax handshake tones. The dynamic modification of session from audio to image is accomplished through SIP RE-INVITE messages. I would imagine canreinvite= flag controls if an end-point is allowed to send/recv RE-INVITE to/from Asterisk. If so, you'll need to set it to yes for T.38 to work. On 2/19/07, *Thomas Deillon* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I make others tests. Analog Fax 1 - PATTON M-ATA - Asterisk - PATTON M-ATA - Analog Fax2 It works only if I use canreinvite= yes. But all my clients are behind a Nat without ALG or stun stuffs... Do you know if canreinvite= yes it's the only way to make it works?? Thanks a lot for your help, Thomas -Message d'origine- De: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] De la part de Thomas Deillon Envoyé: jeudi, 15. février 2007 11:26 À: Asterisk Users Mailing List - Non-Commercial Discussion Objet: [asterisk-users] Fax with T.38 Hi all, I make mistakes in my explanation, so I will try to re-explain my problem… I want to send fax with FoIP. Analog Fax ← PSTN → PATTON SN4960 ←T.38→ Asterisk ←T.38→ PATTON M-ATA ←Analog→ Analog Fax 2 In the Patton SN4960 configuration I have : profile voip FOIP codec 1 g729 rx-length 20 tx-length 20 no-silence-suppression codec 2 g711alaw64k rx-length 30 tx-length 30
[asterisk-users] upgrading from A101 to....A102
Any benefit on getting the PCI Express version? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] iaxmodem - fax tone?
Yes the Sangoma turns off echo cancellation when other faxes come into that DID (I see it in the logs/console) Ironically our office fax is the rxfax :) No analog in sight here! I will test to an efax, I can't believe I forgot about my number there. Asterisk does funny things to the mind when you realize you can do everything yourself! Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU Sent: Friday, February 16, 2007 5:44 PM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] iaxmodem - fax tone? From: Bill Gibbs [EMAIL PROTECTED] Date: Fri, 16 Feb 2007 15:55:13 -0500 I am testing out hylafax and iaxmodem. Submitting jobs to the hylafax server and having it then dial using iaxmodem is working fine. Hylafax server is talking to my Asterisk box that has a Sangoma A101 in it via iaxmodem via an IAX channel using ulaw. A call coming into a certain test DID comes into the Sangoma A101 then it goes to another box via IAX ulaw that uses the rxfax app to capture the fax. I am using rxfax right now because I am just starting to test hylafax. Normal faxes into my test DID work fine so I know it's not the Sangoma or rxfax app or communication between those 2 servers. I notice the Sangoma detects incoming fax tones for these faxes. However, if I send the fax via iaxmodem, I notice the Sangoma sends the call out via my trunk group that turns off the echo cancellation (Because I am using a group I setup to do that, there it's surprise there) but when it comes back in, the Sangoma is not turning off echo cancellation. The end result is rxfax gets the tiff image, but then the call just hangs up and the pdf is never created and sent. The Tiff image appears to be complete however but the call just hangs up after that. I don't have experience with iaxmodem (although your posting just gave me great confidence that I could use it). But you missed one piece of info: when normal FAX' send to your test DID, does Sangoma turn off echo cancellation on that channel? This would be a useful test to confirm your theory that echo cancellation is causing the problem. Another piece of useful information would be, when you say when it comes back in, do you mean you are using iaxmodem to dial another channel in the trunk group it dials out, and that channel uses iaxmodem on the same server to receive the FAX call? Yet another important - and easy - test would be the one you haven't done: send a FAX via the trunk to an alalogue FAX. Or may be you have? I mean, send a FAX via Sangoma to an alalogue line to receive the FAX. If you are concerned about spamming other people's FAX machine, you can even set up eFAX for free in one minute, and send the test FAX to yourself. (But get to wonder how your office FAX is connected.:-) Yuan Liu I think the problem is due to the Sangoma not detecting the fax tones. Am I missing a setting with iaxmodem or hylafax? To recap: Test DID works fine with normal analog and other fax via ATA adapters so I think I can safely rule out a misconfiguration there Iaxmodem registered, hylafax clearly sends the fax via it as I see it coming back in and the tiff created using rxfax The problem appears to be coming back - echo cancellation not being turned off. Unfortunately like many of us, I don't have a test PRI server I can play with so I have to do this after hours. I will be turning off echo cancellation late at night and seeing if that solves the problem but wanted to pose this question to the list. I have _not_ tested it to an outside analog fax yet via hylafax/iaxmodem. Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] iaxmodem - fax tone?
Sending out works great. No problems to external fax machines. So it's something with sending it out and coming back in...where the tone is not being detected. Is there anyway to force the Sangoma to disable echo cancellation on the fly? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU Sent: Friday, February 16, 2007 5:44 PM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] iaxmodem - fax tone? From: Bill Gibbs [EMAIL PROTECTED] Date: Fri, 16 Feb 2007 15:55:13 -0500 I am testing out hylafax and iaxmodem. Submitting jobs to the hylafax server and having it then dial using iaxmodem is working fine. Hylafax server is talking to my Asterisk box that has a Sangoma A101 in it via iaxmodem via an IAX channel using ulaw. A call coming into a certain test DID comes into the Sangoma A101 then it goes to another box via IAX ulaw that uses the rxfax app to capture the fax. I am using rxfax right now because I am just starting to test hylafax. Normal faxes into my test DID work fine so I know it's not the Sangoma or rxfax app or communication between those 2 servers. I notice the Sangoma detects incoming fax tones for these faxes. However, if I send the fax via iaxmodem, I notice the Sangoma sends the call out via my trunk group that turns off the echo cancellation (Because I am using a group I setup to do that, there it's surprise there) but when it comes back in, the Sangoma is not turning off echo cancellation. The end result is rxfax gets the tiff image, but then the call just hangs up and the pdf is never created and sent. The Tiff image appears to be complete however but the call just hangs up after that. I don't have experience with iaxmodem (although your posting just gave me great confidence that I could use it). But you missed one piece of info: when normal FAX' send to your test DID, does Sangoma turn off echo cancellation on that channel? This would be a useful test to confirm your theory that echo cancellation is causing the problem. Another piece of useful information would be, when you say when it comes back in, do you mean you are using iaxmodem to dial another channel in the trunk group it dials out, and that channel uses iaxmodem on the same server to receive the FAX call? Yet another important - and easy - test would be the one you haven't done: send a FAX via the trunk to an alalogue FAX. Or may be you have? I mean, send a FAX via Sangoma to an alalogue line to receive the FAX. If you are concerned about spamming other people's FAX machine, you can even set up eFAX for free in one minute, and send the test FAX to yourself. (But get to wonder how your office FAX is connected.:-) Yuan Liu I think the problem is due to the Sangoma not detecting the fax tones. Am I missing a setting with iaxmodem or hylafax? To recap: Test DID works fine with normal analog and other fax via ATA adapters so I think I can safely rule out a misconfiguration there Iaxmodem registered, hylafax clearly sends the fax via it as I see it coming back in and the tiff created using rxfax The problem appears to be coming back - echo cancellation not being turned off. Unfortunately like many of us, I don't have a test PRI server I can play with so I have to do this after hours. I will be turning off echo cancellation late at night and seeing if that solves the problem but wanted to pose this question to the list. I have _not_ tested it to an outside analog fax yet via hylafax/iaxmodem. Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iaxmodem - fax tone?
I am testing out hylafax and iaxmodem. Submitting jobs to the hylafax server and having it then dial using iaxmodem is working fine. Hylafax server is talking to my Asterisk box that has a Sangoma A101 in it via iaxmodem via an IAX channel using ulaw. A call coming into a certain test DID comes into the Sangoma A101 then it goes to another box via IAX ulaw that uses the rxfax app to capture the fax. I am using rxfax right now because I am just starting to test hylafax. Normal faxes into my test DID work fine so I know it's not the Sangoma or rxfax app or communication between those 2 servers. I notice the Sangoma detects incoming fax tones for these faxes. However, if I send the fax via iaxmodem, I notice the Sangoma sends the call out via my trunk group that turns off the echo cancellation (Because I am using a group I setup to do that, there it's surprise there) but when it comes back in, the Sangoma is not turning off echo cancellation. The end result is rxfax gets the tiff image, but then the call just hangs up and the pdf is never created and sent. The Tiff image appears to be complete however but the call just hangs up after that. I think the problem is due to the Sangoma not detecting the fax tones. Am I missing a setting with iaxmodem or hylafax? To recap: Test DID works fine with normal analog and other fax via ATA adapters so I think I can safely rule out a misconfiguration there Iaxmodem registered, hylafax clearly sends the fax via it as I see it coming back in and the tiff created using rxfax The problem appears to be coming back - echo cancellation not being turned off. Unfortunately like many of us, I don't have a test PRI server I can play with so I have to do this after hours. I will be turning off echo cancellation late at night and seeing if that solves the problem but wanted to pose this question to the list. I have _not_ tested it to an outside analog fax yet via hylafax/iaxmodem. Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bandwidth shapping device
I would use a Mikrotik - www.mikrotik.com Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Wednesday, February 14, 2007 9:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Bandwidth shapping device I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] The High Performance Echo Canceller (HPEC)
Will this work with SIP channels? I get zero echo out the PRI but I do get it occasionally on a LD provider (SIP) we use. The stock * install doesn't appear to be doing anything stopping echo on those channels. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nic Bellamy Sent: Tuesday, February 13, 2007 3:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC) Larry Shields wrote: I recently read about the following new technologies from Digium. Has anyone tried the new HPEC or knows when it will be available? It's out now, and I've tried it - the difference between HPEC and MG2 from trunk is stunning - in situations with bad echo where MG2 can take ten or more seconds to converge to a reasonable degree, HPEC does it in perhaps 300ms - converging on my intake of breath before I say hello, and absolutely no echo after that unless I purposefully go out of my way to screw it up (whistling/blowing into the handpiece for instance - even then, the malfunction is minimal). You can now buy it from the Digium website (US$10 per channel), or if you have an in-warranty Digium card, email through the serial numbers to Digium support and they'll give you a key (this is what I did). You'll need Zaptel 1.2.13 to make it go. It does take quite a bit of CPU though - perhaps 70% more compared to MG2-trunk for the same number of taps from my rough measurements. Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfer - announce - ring
I am running some Polycom phones and have Auto Answer setup(*51 initiates that when you call an extension) With an attended transfer you can take a call, hit transfer, *51extension, announce the call and if the person wants it, complete the transfer, the call is now on speaker at the end. This can surprise people because all of a sudden the call is right there. I know that the attended transfer basically puts the first call on hold, makes another call, then bridges them together with the 2nd push of the transfer button, but is there anyway to have it hang up the speaker part and then _ring_ at the destination? Basically what would happen if transferred the call normally without sending the feature to tell the Polycom to auto answer, but at the same time giving them the ability to announce it over speaker, all in one fluid motion? I know you could place the call on hold, Intercom over then do a blind transfer but like all of us here, we have to deal with people who don't like to push buttons. J I told our people to use Park and then just intercom and the destination can pick it up at their leisure but I thought I'd raise the question. That's what this feature is for after all! So to recap as it works now: Call comes in, Transfer, *51exten, destination Polycom auto answers, Transfer at source and the call is on speaker Is it possible to: Call comes in, Transfer, announce to destination via Auto Answer, then complete the transfer but have it ring at the destination? All using the built in Polycom transfer function? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Buddy list order
I could have sworn I saw a post about this recently but I can't find it so apologies if this is a dupe, but is there anyway to control the order in the Polycom Buddies list? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: why there havn't app_meetme.sofileaboutasterisk1.4.0?
Removed the DEPENDS_app_meetme=ZAPTEL from the file menuselect.makedeps And in menuselect.makeopts I removed the DEPSFAILED line that had meetme in it. It then compiled. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ?? Sent: Thursday, February 01, 2007 9:01 AM To: Asterisk Users Mailing List - No Subject: Re: [asterisk-users] Re: why there havn't app_meetme.sofileaboutasterisk1.4.0? Steven,hello! Thank you so much, but I have installed Zaptel before Asterisk. You have to compile and install Zaptel first, for asterisk to build meetme. -- -- Steven http://www.glimasoutheast.org 李君 [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] asterisk-users@lists.digium.com hi, I install asterisk1.4.0 , when I use the meetme application. The console show that WARNING[9872]: pbx.c:1755 pbx_extension_helper: No application 'Meetme' for extension . I found that there havn't app_meetme.so in the directory of moudles. Then I complied the asterisk1.4.0 again , there is no app_meetme.so also. How to overcome this problem? Thanks, Amy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = = = = = = = = = = = = = = = = = = = = 致 礼！ 李君 [EMAIL PROTECTED] 2007-02-01 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Testing IVR / Callcenter applications
What do you mean? Setup another box, make a bunch of calls (as if you were clients) into the production box, use back to back E1 cards. Bill From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of fadi mujahid Sent: Wednesday, January 31, 2007 10:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Testing IVR / Callcenter applications Thanks for ur suggestion. But the problem is that won't test the queuing of the outbound and inbound calls of the callcenter thanks again On 1/31/07, Alejandro Lengua [EMAIL PROTECTED] wrote: Why don´t you put the IVR in an extension... and call it also from an extension of the same PBX. On 1/31/07, fadi mujahid [EMAIL PROTECTED] wrote: Hello We are developing an application to be deployed on E1 lines (inbound and outbound calls) What is the best way to fully test the application if we do not have E1 lines in the development environment? Is there some kind of software tester to test IVR/Callcenter applications virtually?? thanks and best regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] No intercom splash tone?
If you have r option in the Dial command remove it. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken Morley Sent: Tuesday, January 30, 2007 9:06 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] No intercom splash tone? Environment: Asterisk 1.2.14, FreePBX 2.2.0, Aastra 480i IP telephones firmware version 1.4.1.1077. Problem: Intercom feature: the dialed phone does not play the splash tone when auto-answering an intercom call. Otherwise, intercom works perfectly. Questions: What is the extensions.conf syntax to trigger a splash tone in Asterisk 1.2.14 (from the documentation and posts I've found, it has changed as Asterisk has evolved)? What SIP header, etc. does the Aastra 480i firmware 1.4.1.1077 need to play a splash tone? What is the suggested methodology for further troubleshooting? Many thanks! Ken Morley ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] web-meetme cbmysql not registered
I am experiencing the same problem. Fresh install. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ma Zhiyong Sent: Tuesday, January 30, 2007 6:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] web-meetme cbmysql not registered HI, today I download Web-MeetMe-3.0.0 for asterisk 1.4.0 but when I call the extension which invoke cbmysql, a warning appears: WARNING[20225] pbx.c: No application 'CBMysql' for extension (default, 1995, 3) I check the application, it didn't registered CLI core show application CBMySQL Your application(s) is (are) not registered But I can see it use show module and in my start log, it shows [Jan 30 18:40:15] VERBOSE[6702] logger.c: == Parsing '/etc/asterisk/cbmysql.conf': [Jan 30 18:40:15] VERBOSE[6702] logger.c: Found [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: got hostname of localhost [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: got port of 3306 [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: got sock file of /var/lib/mysql/mysql.sock [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: got user of root [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: got dbname of meetme [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: got password of [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: Using Database for Admin User Options [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: got Connference Application of MeetMe [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: got Conference Count Application of MeetMeCount [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMuSQL: Early Alert set to 300 seconds. [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: Successfully connected to MySQL database. this seems it was loaded successful. what's the matter? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] web-meetme cbmysql not registered
-pstngw1-3 [Jan 30 12:27:11] DEBUG[3619] devicestate.c: No provider found, checking channel drivers for IAX2 - trunk-pstngw1 [Jan 30 12:27:11] DEBUG[3619] chan_iax2.c: Checking device state for device trunk-pstngw1 [Jan 30 12:27:11] DEBUG[3619] chan_iax2.c: iax2_devicestate: Found peer. What's device state of trunk-pstngw1? addr=54609872, defaddr=0 maxms=2000, lastms=3 [Jan 30 12:27:11] DEBUG[3619] devicestate.c: Changing state for IAX2/trunk-pstngw1 - state 1 (Not in use) -Original Message- From: [EMAIL PROTECTED] on behalf of Bill Gibbs Sent: Tue 1/30/2007 12:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] web-meetme cbmysql not registered I am experiencing the same problem. Fresh install. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ma Zhiyong Sent: Tuesday, January 30, 2007 6:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] web-meetme cbmysql not registered HI, today I download Web-MeetMe-3.0.0 for asterisk 1.4.0 but when I call the extension which invoke cbmysql, a warning appears: WARNING[20225] pbx.c: No application 'CBMysql' for extension (default, 1995, 3) I check the application, it didn't registered CLI core show application CBMySQL Your application(s) is (are) not registered But I can see it use show module and in my start log, it shows [Jan 30 18:40:15] VERBOSE[6702] logger.c: == Parsing '/etc/asterisk/cbmysql.conf': [Jan 30 18:40:15] VERBOSE[6702] logger.c: Found [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: got hostname of localhost [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: got port of 3306 [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: got sock file of /var/lib/mysql/mysql.sock [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: got user of root [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: got dbname of meetme [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: got password of [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: Using Database for Admin User Options [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: got Connference Application of MeetMe [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: got Conference Count Application of MeetMeCount [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMuSQL: Early Alert set to 300 seconds. [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: Successfully connected to MySQL database. this seems it was loaded successful. what's the matter? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 1.4 problem with ztdummy and MeetMe()
Here is what he was getting at: 1.2.x and ztdummy and meetme all work fine. However Compile Zaptel 1.4.0, install, reboot Zttool shows ztdummy as the timing device. Lsmod shows it loaded. If you then compile Asterisk 1.4.0 it fails to compile app_meetme My quick and dirty solution just now: Removed the DEPENDS_app_meetme=ZAPTEL from the file menuselect.makedeps And in menuselect.makeopts I removed the DEPSFAILED line that had meetme in it. It then compiled. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oded Arbel Sent: Sunday, January 28, 2007 5:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 problem with ztdummy and MeetMe() On Thu, 2007-01-25 at 18:40 +0100, Stefan Wintermeyer wrote: Hi, when I build zaptel-1.2 and asterisk-1.2 I can modprobe ztdummy and start asterisk to be able to use MeetMe(). When I build zaptel-1.4 and asterisk-1.4 I can modprobe ztdummy and start asterisk but I am not able to use MeetMe(). What do I miss? I'm not sure, because we missed the entire problem description, which I would imaging would have included log snippets and/or error message reports, but it was apparently removed from your e-mail. http://www.catb.org/~esr/faqs/smart-questions.html -- Oded Arbel Atelis [EMAIL PROTECTED] Tel: +972-54-7340014 ::.. He's dead, Jim. You grab his wallet, I'll grab his tricorder. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] web-meetme cbmysql not registered
That solved the problem thank you. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Austin Sent: Tuesday, January 30, 2007 1:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] web-meetme cbmysql not registered Ma wrote: WARNING[20225] pbx.c: No application 'CBMysql' for extension (default, 1995, 3) I check the application, it didn't registered CLI core show application CBMySQL Your application(s) is (are) not registered But I can see it use show module I made a small mess of supporting the new module loading process. The code attempts to determine if the config file was successfully loaded, and only then load the module and register the application. That is all fine and well, except I failed to properly flag a successful config load. How it ever worked for me, I don't know, but here is a quick fix: Find this section of the code- ast_log(LOG_NOTICE,Successfully connected to MySQL database.\n); connected = 1; records = 0; connect_time = time(NULL); } And add this: if (connected) return 1; else return 0; I'll get an update into svn if this works for you and release 3.0.1 Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4 - SLA
I have read that 1.4 has shared line appearances, which I assume will work with Polycom phones. Has anyone configured this and verified it working? I was going to start playing around with it but wanted to see if anyone else has tackled it yet. Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 1.4 - SLA
Thanks, I had a notebook crash and must have missed that. Appreciate the replies! I will be patient. Bill -Original Message- From: [EMAIL PROTECTED] on behalf of Steve Langstaff Sent: Thu 1/25/2007 9:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] 1.4 - SLA On Fri, 12 Jan 2007 07:14:08 -0800 Kevin P. Fleming wrote: I won't waste your time, because the current SLA implementation is broken. We expect to have replaced it when Asterisk 1.4.1 is released, and there will be better documentation at that point as well. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: 25 January 2007 14:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] 1.4 - SLA I have read that 1.4 has shared line appearances, which I assume will work with Polycom phones. Has anyone configured this and verified it working? I was going to start playing around with it but wanted to see if anyone else has tackled it yet. Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] RE: Polycom buddies question
Hints in extensions.conf in conjuction with mac-directory.xml with bw set to 1. Bill -Original Message- From: [EMAIL PROTECTED] on behalf of Doug Sent: Thu 1/18/2007 1:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: Polycom buddies question At 11:56 1/18/2007, Bill Gibbs wrote: Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary=_=_NextPart_001_01C73B2A.03C9AD84 A follow up (late better than never) Finally had time to sit down and look at this sip.cfg keys key.scrolling.timeout=1 key.IP_500.31.function.prim=BuddyStatus/ This turns the Services key which I never use on a 501 into the Buddy Status. It even works while on a call. One touch! How do you know which buddy is being monitored? Does this show a screen of buddies? Bill -- From: Bill Gibbs Sent: Thursday, December 07, 2006 2:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: Polycom buddies question Figures I email this and realized I can hit Menu 1 (Features) 4 (Presence) 2 (Buddy Status) Wow that's a lot of key strokes. Anyway to reduce that to a one button touch? I don't mind doing that but I guess I should think of the users J Bill -- From: Bill Gibbs Sent: Thursday, December 07, 2006 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Polycom buddies question I know this is not asterisk specific but we all know this group is used for Polycom issues as well. I have hints working ok on Asterisk. However the Polycom phone will only show the buddies key if there is not a call. This defeats the purpose of using the buddies to see if you can transfer a call to another extension (using the Buddy key to see if they are on the phone). Polycom sip version: 1.6.6.0036 for all platforms except 11402_001 1.6.6.0042 for 11402_001 Any way around this? The big issue is moving from a key system to this is the ability to use the phone to see who is on the phone so you know if you can transfer a call. Obviously web based interfaces work but that draws your attention from the phone to the computer reducing effectiveness. Buddies half solve this. Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] RE: Polycom buddies question
yes it shows the normal Buddies screen that is available from the LCD if that feature is enabled in the Polycom sip config file (presence) Bill -Original Message- From: [EMAIL PROTECTED] on behalf of Doug Sent: Thu 1/18/2007 1:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: Polycom buddies question At 11:56 1/18/2007, Bill Gibbs wrote: Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary=_=_NextPart_001_01C73B2A.03C9AD84 A follow up (late better than never) Finally had time to sit down and look at this sip.cfg keys key.scrolling.timeout=1 key.IP_500.31.function.prim=BuddyStatus/ This turns the Services key which I never use on a 501 into the Buddy Status. It even works while on a call. One touch! How do you know which buddy is being monitored? Does this show a screen of buddies? Bill -- From: Bill Gibbs Sent: Thursday, December 07, 2006 2:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: Polycom buddies question Figures I email this and realized I can hit Menu 1 (Features) 4 (Presence) 2 (Buddy Status) Wow that's a lot of key strokes. Anyway to reduce that to a one button touch? I don't mind doing that but I guess I should think of the users J Bill -- From: Bill Gibbs Sent: Thursday, December 07, 2006 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Polycom buddies question I know this is not asterisk specific but we all know this group is used for Polycom issues as well. I have hints working ok on Asterisk. However the Polycom phone will only show the buddies key if there is not a call. This defeats the purpose of using the buddies to see if you can transfer a call to another extension (using the Buddy key to see if they are on the phone). Polycom sip version: 1.6.6.0036 for all platforms except 11402_001 1.6.6.0042 for 11402_001 Any way around this? The big issue is moving from a key system to this is the ability to use the phone to see who is on the phone so you know if you can transfer a call. Obviously web based interfaces work but that draws your attention from the phone to the computer reducing effectiveness. Buddies half solve this. Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Polycom buddies question
A follow up (late better than never) Finally had time to sit down and look at this sip.cfg keys key.scrolling.timeout=1 key.IP_500.31.function.prim=BuddyStatus/ This turns the Services key which I never use on a 501 into the Buddy Status. It even works while on a call. One touch! Bill From: Bill Gibbs Sent: Thursday, December 07, 2006 2:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: Polycom buddies question Figures I email this and realized I can hit Menu 1 (Features) 4 (Presence) 2 (Buddy Status) Wow that's a lot of key strokes. Anyway to reduce that to a one button touch? I don't mind doing that but I guess I should think of the users :-) Bill From: Bill Gibbs Sent: Thursday, December 07, 2006 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Polycom buddies question I know this is not asterisk specific but we all know this group is used for Polycom issues as well... I have hints working ok on Asterisk. However the Polycom phone will only show the buddies key if there is not a call. This defeats the purpose of using the buddies to see if you can transfer a call to another extension (using the Buddy key to see if they are on the phone). Polycom sip version: 1.6.6.0036 for all platforms except 11402_001 1.6.6.0042 for 11402_001 Any way around this? The big issue is moving from a key system to this is the ability to use the phone to see who is on the phone so you know if you can transfer a call. Obviously web based interfaces work but that draws your attention from the phone to the computer reducing effectiveness. Buddies half solve this... Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fax through Sangoma A102
Incoming faxes, the Sangoma will detect the tones and disable echo cancel. To send outbound, you will have to add another trunk group, of one or more channels and disable echo cancellation and use that to dial out. Example (/etc/asterisk/zapata.conf) blah blah echocancel=yes blah blah group = 1 channel =1-20 blah blah echo cancel=no group = 2 channel=22-23 So you would use for faxing specifically Zap/g2/number and it will use channel 22 or 23 but with echo turned off. Use Zap/g1 and it will use the first group of channels with echo cancel on (or whatever other parameters come before the group command) Bill From: [EMAIL PROTECTED] on behalf of jeremij jerome Sent: Tue 1/9/2007 10:52 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Fax through Sangoma A102 Hello, in our company we are trying to do this: Fax -- Traditional PBX -- Asterisk -- PSTN In practice, we have put an Asterisk equipped with a Sangoma A102 (2 PRI ports) between our PBX (Siemens HiCom) and the PSTN in order to have a VoIP network along the traditional telephony network. The problem is with the fax. We just want to send and receive faxes from/to our fax machine connected to the Siemens (without needing any interaction with our VoIP network, the faxes are sent to/received from PSTN). Unfortunately we are experiencing a lot of problems: the faxes not always work and when they work, it's likely to have incomplete pages. I know that faxing with VoIP is very troublesome, but maybe someone else is using a similar configuration and he found a good configuration or maybe has some hints to improve the results. We are using Asterisk 1.2.13. Thanks, Jeremi ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] yet another faxing issue (outbound only, via ATA)
I set aside a couple of channels and removed echo cancellation on them. So far, faxing outbound through an ATA is working fine now. Bill -Original Message- From: [EMAIL PROTECTED] on behalf of Marco Mouta Sent: Wed 1/3/2007 6:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] yet another faxing issue (outbound only, via ATA) Hi all, I was having a similar issue, using TE110P from Digium all incoming faxes were detected and correctly received. When trying to send outbound faxes, they all get broken... I do believe it may be related with Echo Cancel enabled on my Zapata.conf, any ways i've set also fax detect for inbound and outbound on zapata, but that stills diferent from the receiving model as it relies on NVfaxdetect to detect. After many trials, i setup an architecture with another Server Running Hylafax and IAXmodem registring on my * Box and i just get out of troubles. It's perfect sending and receiving faxes with notifications and everything else, Hylafax + IAXmodem and Asterisk are working like a charm. I must say that we don't send too many faxes per day, but until now no problems! And yes didn't change anything on Zapata config or something else on Asterisk Box, i just added IAX account registred my Hylafax IAXmodem there and Voilá :) On 1/3/07, Thomas Kenyon [EMAIL PROTECTED] wrote: Bill Gibbs wrote: My next step is to connect the fax machine to a Wildcard X100P. Check to see if there is Echo cancellation in the SPA-1001, and if so turn it off. If there is an adaptive Jitter buffer on the SPA-1001, try changing it to a fixed one (probably no more than 40ms). Why would you connect a fax machine to an X100P, aren't they FXO cards? Have you tried terminating to a VOIP provider? (to see if the problem is with the ATA). Here I use a fax machine connected to a CS6220 which is connected to the asterisk box and terminates with a TDM400P card (so a completely different arrangement). Any other suggestions? Black magic? Voodoo? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Detect IP path before calling
If you send the SIP call to the remote end which is no longer available (unreachable, etc) and have another Dial statement, it will automatically roll over. I would think this would be just as fast, if not faster, than a script updating a db value you check before each call. Bill -Original Message- From: [EMAIL PROTECTED] on behalf of Yuan LIU Sent: Wed 1/3/2007 10:33 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Detect IP path before calling From: Paul Hales [EMAIL PROTECTED] With the chanisavail command. PaulH Doesn't seem to have effect. Probably I should state the problem more clearly. Ideally, Asterisk should not attempt SIP if there's no way to establish a SIP call. This may include lack of IP connection (ping timeout, for example), or no SIP listener on remote side (this would be difficult because Asterisk can only use UDP). My environment does not require remote end point to register, so consulting the registry is not an option. (This is perhaps what ChanIsAvail does.) Any suggestions? I'll go to scripting if no other easy way. Yuan Liu On Wed, 2007-01-03 at 14:22 -0800, Yuan LIU wrote: Any easy way to determine if IP connectivity before attempting a SIP call? IP connectivity could be a timeout. Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] yet another faxing issue (outbound only, via ATA)
2 Asterisk servers 1.2.12.1 Connected via IAX2, same switch, GigE, no packet loss, etc 1 with a Sangoma A101 for a PRI to the PSTN Ulaw QoS enabled NAT for the registered ATA boxes, no nat between the * servers Faxing inbound: Call from PRI hits the first Asterisk server Then talks to the 2nd via IAX2 NVFaxDetect receives the fax, converts to PDF and emails it out Works great! Never had an issue The problem, however is outbound. Sipura 1001 ATAs. Fax machine connected to the ATA. Registered to the 2nd asterisk box. Keep in mind this server runs voice calls just fine. Outbound calls from this box are ulaw The call is then sent via IAX2 (also tried SIP as well) to the Asterisk server w/ the PRI, then out to the world Hit and miss to send faxes out Echo cancellation is enabled on the PRI I have lowered the rxgain and txgain to -5.0, seems fine for voice. The ATA is running 3.1.8 firmware from Sipura with fax detect turned Usually the faxes fail, but sometimes you will get all the pages, but only a fraction of the page. I have tried turning off ECM but still the same issue. I would suspect the Sangom or IAX2, or something of that nature except receiving faxes traveling to the 2nd asterisk box works just fine! I also tried to register the ATA to the primary Asterisk server w/ the PRI, same exact issue. Any ideas - better luck w/ Grandstream? I suspect the problem is not Asterisk, or the Sangoma, or jitter or bandwidth since receiving faxes works fine. I did not try to receive faxes through the ATA to the machine itself, I tried that a few months ago during other testing, never got it to work so I never tried again once I got NVFaxDetect working for email. My next step is to connect the fax machine to a Wildcard X100P. Any other suggestions? Black magic? Voodoo? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: yet another faxing issue (outbound only, via ATA)
Follow up: I used my Cisco 3660 that's a hop away and connected to a different PRI provider. Faxes work _fine_ From the ATA box I faxed a DID that would come back into the Zap enabled Asterisk server, then talks BACK to the server that the ATA box is regstered via IAX2 (or SIP, I found they both worked) and was able to receive the fax fine (incoming fax went to email) FAILURE FAX: Here is the path: ATA - SIP - * - IAX2 (or SIP) - * with Zap - send call out via Zap channel SUCCESS FAX: ATA - SIP - * - IAX2 (or SIP) - * with Zap - SIP to 3660 then out via PRI works every time! I tried G3 and ECM mode. ECM was flakey work even through the 3660 but G3 worked everytime. I have set the fax machine to G3 for the time being since it works each time. Each outbound call actually initiates a call to a DID that terminates into my * with the Zap card, then talks via IAX2 again back to the original server. No problems there. So I know that faxing other fax machines fails so it's not necessarily that there is some weird loop calling out and coming back in the same Zap card is there? Recap hardware: Sangoma A101 Echo cancel: yes (and zap show channel confirms it's enabled) I would think if echo cancel was the problem incoming faxes would fail as well? So...wtf?! I am surprised the 3660 is working outbound where the Sangoma is not, since it can receive fine. The 3660 has a HDV card in it with DSPs to do the processing but the load on the Asterisk servers barely goes above 0.00. So to recap: ATA works fine sending and my Asterisk servers are ok Sending the outbound call via SIP to my 3660 a hop away (DS3) to be routed out the PSTN (which then comes back to my Asterisk with the Sangoma card) works fine! Sending the outbound fax via the Zap channels on the Asterisk server (the same one that talks to the 3660 via SIP that works) FAILS Receiving faxes from anywhere into the Sangoma which talks to my 2nd asterisk server works fine as well! Bill -Original Message- From: Bill Gibbs Sent: Tue 1/2/2007 2:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: yet another faxing issue (outbound only, via ATA) 2 Asterisk servers 1.2.12.1 Connected via IAX2, same switch, GigE, no packet loss, etc 1 with a Sangoma A101 for a PRI to the PSTN Ulaw QoS enabled NAT for the registered ATA boxes, no nat between the * servers Faxing inbound: Call from PRI hits the first Asterisk server Then talks to the 2nd via IAX2 NVFaxDetect receives the fax, converts to PDF and emails it out Works great! Never had an issue The problem, however is outbound. Sipura 1001 ATAs. Fax machine connected to the ATA. Registered to the 2nd asterisk box. Keep in mind this server runs voice calls just fine. Outbound calls from this box are ulaw The call is then sent via IAX2 (also tried SIP as well) to the Asterisk server w/ the PRI, then out to the world Hit and miss to send faxes out Echo cancellation is enabled on the PRI I have lowered the rxgain and txgain to -5.0, seems fine for voice. The ATA is running 3.1.8 firmware from Sipura with fax detect turned Usually the faxes fail, but sometimes you will get all the pages, but only a fraction of the page. I have tried turning off ECM but still the same issue. I would suspect the Sangom or IAX2, or something of that nature except receiving faxes traveling to the 2nd asterisk box works just fine! I also tried to register the ATA to the primary Asterisk server w/ the PRI, same exact issue. Any ideas - better luck w/ Grandstream? I suspect the problem is not Asterisk, or the Sangoma, or jitter or bandwidth since receiving faxes works fine. I did not try to receive faxes through the ATA to the machine itself, I tried that a few months ago during other testing, never got it to work so I never tried again once I got NVFaxDetect working for email. My next step is to connect the fax machine to a Wildcard X100P. Any other suggestions? Black magic? Voodoo? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] RE: yet another faxing issue (outbound only, via ATA)
Haven't yet. Gotta wait until the calls stop flowing in/out. It's a production system. That's on the list of tonight. Bill From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora Sent: Tuesday, January 02, 2007 8:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: yet another faxing issue (outbound only,via ATA) On 1/2/07, Bill Gibbs [EMAIL PROTECTED] wrote: Echo cancel: yes (and zap show channel confirms it's enabled) I would think if echo cancel was the problem incoming faxes would fail as well? This is only a guess. The Sangoma is detecting the fax when it receives it, and is turning off echo cancel. However, when box b sends via IAX2 or SIP to box a, the Sangoma no longer knows that it is a fax transmission and is continuing echo cancellation. The Cisco 3660 recognizes that it is a fax and turns off echo (or doesn't have echo cancellation). Question: If you turn OFF echo cancellation, does it work then? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re:Re: outgoing call on ISDN PRI
Talk to your carrier. Most likely you won't be able to hide it. You
might be able to set it.
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michel
Sent: Wednesday, December 13, 2006 7:56 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re:Re: outgoing call on ISDN PRI
Sorry, sorry !!! I was mixed with another config when I wrote my first
email !!
In fact, User A is registered on Asterisk and user B has a public phone
number (no link with Asterisk).
Our test is : User A calls asterisk server via SIP. As User A context
has a DIAL('user B phone number'),
Asterisk calls user B via ISDN line. Then, user B phone rings and we
can see the caller phone number on
user B phone screen. This caller number is our ISDN line number. What we
would like to do is to hide the caller number (our ISDN line number).
We tried usecallerid, callerid, hidecallerid, restrictcid,
usecallingpres in zapata.conf but it doesn't work.
Do you or anyone know how to hide it?
Thanks you!
--
Message: 4
Date: Tue, 12 Dec 2006 19:04:44 +
From: Tim Panton [EMAIL PROTECTED]
Subject: Re: [asterisk-users] outgoing call on ISDN PRI
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed
On 12 Dec 2006, at 15:11, Michel wrote:
HEllo list !
When user A calls user B via Asterisk (Users A and B are registered
on the same Asterisk server ) and an ISDN PRI, user B phone
always shows Asterisk server telephone number. How to hide it and
how to forward user A number ?
We tried usecallerid, callerid, hidecallerid, restrictcid,
usecallingpres in zapata.conf but we always see Asterisk server
telephone number !
I'm not getting a clear picture of how the ISDN PRI gets into it if
both users are registered (SIP I assume)
to the same asterisk.
If the call actually goes out via a Public ISDN line, you have to get
the provider to agree to let
you set the outgoing number. Normally they will only let you set it
to one of the inbound numbers
that you have bought from them :-)
If that doesn't help,
please re-phrase the question...
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial groups, groups of phones, multiple line keys
I have 4 Polycom phones with multiple line keys so multiple incoming calls work fine The way I would like the incoming call flow to work is as follows: 1) 2 groups consisting of 2 phones each 2) Incoming call rings the first group, if no answer, the 2nd group is rung 3) However if the first 2 are on a call or busy, it will immediately ring the 2nd group 4) If one of the first group is in use, the available phone is rung, if no answer, roll over to group 2 5) If group 2 one phone is busy, ring the other one only 6) Finally drop into voicemail if no answer at all Suggestions on how to do that yet still keep the multiple line keys? Would this be a good use of CheckGroup and Set(GROUP())? I could use astdb but I wanted to stay away from persistent variables. I looked into ChanIsAvail but I don't think that is what I want. So I guess what I am looking for is there a way to find out if a device is using ANY channel, because I can check that (say CheckIfPhoneInUse(SIP/phone1)) and set dynamic variable values based on that, then decide what phones to ring based on that. Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom buddies question
I know this is not asterisk specific but we all know this group is used for Polycom issues as well... I have hints working ok on Asterisk. However the Polycom phone will only show the buddies key if there is not a call. This defeats the purpose of using the buddies to see if you can transfer a call to another extension (using the Buddy key to see if they are on the phone). Polycom sip version: 1.6.6.0036 for all platforms except 11402_001 1.6.6.0042 for 11402_001 Any way around this? The big issue is moving from a key system to this is the ability to use the phone to see who is on the phone so you know if you can transfer a call. Obviously web based interfaces work but that draws your attention from the phone to the computer reducing effectiveness. Buddies half solve this... Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Polycom buddies question
Figures I email this and realized I can hit Menu 1 (Features) 4 (Presence) 2 (Buddy Status) Wow that's a lot of key strokes. Anyway to reduce that to a one button touch? I don't mind doing that but I guess I should think of the users :-) Bill From: Bill Gibbs Sent: Thursday, December 07, 2006 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Polycom buddies question I know this is not asterisk specific but we all know this group is used for Polycom issues as well... I have hints working ok on Asterisk. However the Polycom phone will only show the buddies key if there is not a call. This defeats the purpose of using the buddies to see if you can transfer a call to another extension (using the Buddy key to see if they are on the phone). Polycom sip version: 1.6.6.0036 for all platforms except 11402_001 1.6.6.0042 for 11402_001 Any way around this? The big issue is moving from a key system to this is the ability to use the phone to see who is on the phone so you know if you can transfer a call. Obviously web based interfaces work but that draws your attention from the phone to the computer reducing effectiveness. Buddies half solve this... Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Sticky Polycom 501 keys and handset
I think in the features you can completely wipe the sip image Menus Settings Advanced Admin Settings Reset to default Then format the file system Bill From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Tuesday, November 07, 2006 1:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Sticky Polycom 501 keys and handset Any hints on downgrading? I placed the old SIP 1.6.7 on the right folder, but my phone wont pick it up and install it. It must be thinking this is an old version, ignore or something I`ve never downgraded a phone, I tend to like upgrading more :-) Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rick Smith Sent: November 7, 2006 11:28 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Sticky Polycom 501 keys and handset I had this EXACT same problem, and 2.0.x is the problem according to Polycom Tech Support. I had such a hard time explaining the problem, too Downgraded to 1.6.7 and all worked well again. Polycom says if youre using Asterisk, dont go past 1.6.7 until they say to. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Tuesday, November 07, 2006 11:02 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Sticky Polycom 501 keys and handset Hi, I've recently bought new Polycom 501 phones, upgraded to bootrom 3.2.2 and SIP 2.0.1. I just noticed something, which I first blamed on Asterisk and NATs (a 2 second silence at the beginning of a call). Something I'venoticed also on my old phone (which is having the same problem now, but its also been upgraded). My keys are sticky. Simple as that. Sometimes I press a number and the key comes up (the hardware seems fine) but the phone produces this lng tone as if I had pressed the key for 3 seconds. Even the receiver is sticky, giving my dialtone when I lift it only1-2 seconds after I lift the handset. It simply looks like the phone can't keep up, like a sluggishcomputer. Anybody has ever seem this? I'd like to downgrde to SIP 1.6.7 to see if the new sip app was the problem. How can I do that? I've placed the old sip.ld file where I had to, but the phone wont pick it up. Short of that, can somebody point me to the newest firmware (2.0.2) to see if thatwould help? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] question about IF
I am having a problem getting the following logic to work,
in a macro.
Basically, if the caller ID matches, set the outbond trunk
to a Zap channel, otherwise use a SIP provider.
exten = s,n,Set(TRUNK=${IF($[${CALLERIDNUM} = 1234567890]?Zap/g1:SIP/LDPROVIDER)})
; use PRI instead of SIP
That works. The TRUNK variable is set properly.
But the SIP LD provider requires a prepended code, so I say
if the TRUNK var is the SIP/LDPROVIDER set another variable called PREPEND_CODE
to the passed argument, otherwise use nothing if its the Zap channel
since I will just be sending the digits as is.
exten = s,n,Set(PREPEND_CODE=${IF($[ ${TRUNK} = SIP/LDPROVIDER]?${ARG2}:)})
However this always sets the PREPEND_CODE variable even if
TRUNK is set to Zap/g1. If I use SIP/LDPROVIDER or SIP/LDPROVIDER
or even ${TRUNK:} = SIP (not sure if thats even valid but I tried it) it
still sets the variable. I also tried using a space after the : but still doesnt
work.
Does IF only match digits???
Asterisk ver 1.2.12.1
I must be doing something wrong but am not sure what it isany
ideas? I have a feeling I missed up because I was looking at this at 2am and
working on it via a cell phone SSH connection so I must be missing something
obvious.
Bill
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] OT: Polycom time sync - sorta
I haven't but how about adjusting the offset by 11 secs to compensate? Lame I know but then you can go back to counting bricks on the side walk again! j/k :) Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton Sent: Thursday, October 05, 2006 2:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] OT: Polycom time sync - sorta Greetings I have a couple polycom phones (501 and 601) I'm messing around with and I've noticed something weird. Both phones synchronize their clocks to a central NTP server here on our network and both phones are 11 seconds slow. All of our servers, switches, routers and PCs also sync to this time source and are spot on. Even the budgetone 101 is spot on. Has anyone else experienced this? I know I'm being anal retentive but it's driving me nuts. The phone is getting it's sntp server and offset settings via DHCP and they show correct on the phone. The phone is running v1.6.7 firmware. Thanks -Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk manager
www.freepbx.org Bill From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Voipers Portugal Sent: Tuesday, October 03, 2006 8:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk manager Dear all, Do you know any tool that can administrate Asterisk remotely? I only need basic functionalities like adding new extensions, queus and basic configuration. The problem is that I can't install that in the same machine as Asterisk (since it is running in open wrt). Can anyone help me out? Jose Simoes ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom 501 One-way Audio
Check the handset cords. They can get loose and cause this exact issue. Bill From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Dugas Sent: Thursday, September 28, 2006 10:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom 501 One-way Audio I have a site running an up-to-date version of Asterisk from the 1.2 trunk. We have a dozen Polycom 501 units and one of them (none of the others) is having recurring one-way-audio problems. As Murphy's Law dictates, it's the bosses phone! The user gets a few calls a day where the caller can hear her fine but she hears dead silence. It happens when she calls out sometimes too. Even internal voicemail and extension-to-extension calls are affected. I just called her three times from another extension; the first two were affected, the third got through. None of the other units seem to have the problem. They're all running the same firmware and are loading central configs that are identical except for line-button text and registration info. I've been running * with lots of debug/verbose logging enabled and have yet to see it complain about anything when she reports the problem. I'm about to replace her phone with a spare to see if that fixes it. Wondering if anyone has seen something like this and might be able to tell me what to look for as a potential cause. TIA, Paul Paul Dugas Computer Engineer Dugas Enterprises, LLC 522 Black Canyon Park Canton, GA 30114 phone: 404.932.1355 fax: 866.751.6494 [EMAIL PROTECTED] http://DugasEnterprises.com This e-mail and any attachments are confidential. If you receive this message in error or are not the intended recipient, you should not retain, distribute, disclose or use any of this information and you should destroy the e-mail and any attachments or copies. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Forcing Transcode
Sure in their sip definition disallow=all allow=g729 Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mr. Jones Sent: Thursday, September 28, 2006 5:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Forcing Transcode Hi Folks, I'm curious if there's anyway to force Asterisk to transcode for certain handsets. Specifically we have an inbound SIP origination service which uses g711. We're having bandwidth issues with a client and would like to force Asterisk to transcode to g729 until we can get their T1 in place. Any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 64 analog phones
I would think channel banks - T1s - TDM card in asterisk server would work better than a bazillion ata adapaters Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Artner Sent: Wednesday, September 27, 2006 8:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 64 analog phones It depends on the actual given environment, but you could also think about using Linksys' PAP2 adapter! mike wrote: Dear list which hardware solution would you suggest for connecting 60 analog phones to asterisk ? thank you very much .mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Dual core
I have a few dual core that I have installed Asterisk on without any issues. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Parcina Sent: Friday, September 22, 2006 3:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dual core Hi list. I have one quick question. Does Asterisk work with dual core processors in version 1.2? Will it work with dual core processors in 1.4? I'm planning to buy new machine for one installation and I have to decide will I buy single or dual core processor. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Looped message playback
Why not just merge the file together a few times using an audio program and make a longer file? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Earle Clubb Sent: Thursday, September 21, 2006 9:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Looped message playback Hello, I'm trying to play an audio file to a phone an arbitrary number of times. The audio is a five-second segment of a sine wave. I need this to be played repeatedly without gaps between playbacks. I've tried doing this in the dial plan, e.g.: exten = s,1,Playback(tonefile) exten = s,2,Goto(1) but there is too long of a gap between the playbacks. Does anyone know of a way to achieve this? Thanks, Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Looped message playback
DOH I just read that you said an arbitrary number of times no wonder you asked this question Please ignore me. :) -Original Message- From: Bill Gibbs Sent: Thursday, September 21, 2006 10:11 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Looped message playback Why not just merge the file together a few times using an audio program and make a longer file? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Earle Clubb Sent: Thursday, September 21, 2006 9:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Looped message playback Hello, I'm trying to play an audio file to a phone an arbitrary number of times. The audio is a five-second segment of a sine wave. I need this to be played repeatedly without gaps between playbacks. I've tried doing this in the dial plan, e.g.: exten = s,1,Playback(tonefile) exten = s,2,Goto(1) but there is too long of a gap between the playbacks. Does anyone know of a way to achieve this? Thanks, Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf for talking to other Asterisk machines
Just curious how most of you are defining SIP peers in sip.conf for Asterisk boxes talking to each other. Are most of you just making a type=friend connection and a single context or are you separating them out to in/out definitions and contexts? In other words Where voicegw1 is the Asterisk box with the TDM cards for talking to the PSTN, it will receive calls from the PSTN and forward to the appropriate Asterisk box as well as receive calls from the other Asterisk boxes to forward out to the PSTN. Do you on the Asterisk box that contains all the SIP phones define (ie the client to the PSTN Asterisk box and voicegw1 is the one with the PSTN connection) [voicegw1-in] type=user username=virtualpbx1-in secret=1234 host=192.168.1.99 context=voicegw1-in canreinvite=no nat=no qualify=yes allow=all [voicegw1-out] type=peer username=virtualpbx1-out secret=1234 host=192.168.1.99 context=voicegw1-out canreinvite=no nat=no qualify=yes allow=all or [voicegw1] Type=friend Blah Context=voicegw1 And use a single context for inbound/outbound routing? The same would apply to the PSTN Asterisk server. Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] saved.gsm - Voicemail greeting ??
Example for mailbox 100 under context default /var/spool/asterisk/voicemail/default/100 -rwx-w 1 asterisk asterisk 442604 Aug 22 16:44 busy.wav -rwx-w 1 asterisk asterisk 44976 Aug 22 16:44 busy.WAV -rwx-w 1 asterisk asterisk 25964 Aug 8 02:17 greet.wav -rwx-w 1 asterisk asterisk 2660 Aug 8 02:17 greet.WAV drwx-w 2 asterisk asterisk 4096 Sep 13 13:45 INBOX drwx-w 2 asterisk asterisk 4096 Sep 13 13:37 tmp -rwx-w 1 asterisk asterisk 168044 Aug 9 17:00 unavail.wav -rwx-w 1 asterisk asterisk 17090 Aug 9 17:00 unavail.WAV drwx-w 2 asterisk asterisk 4096 Sep 1 11:31 Work Those are the files (wav format) that it expects for the voicemail greetings/name announcement. Greet.wav is the name. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Ellson Sent: Friday, September 15, 2006 10:26 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] saved.gsm - Voicemail greeting ?? Hi John, Yes, I followed an example that put all my family sound files in /var/lib/asterisk/sounds/local, which is also where this file is. Now I am trying to figure out how to get the unavailable|name|Busy .gsm's I made loaded into a mailbox without playing my sounds back into a phone ;) Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Fri, 15 Sep 2006, John covici wrote: Check in /var/spool/asterisk/voicemail/default/extension number for a particular extension, don't know how you want to differentiate after hours, etc. Also, you can put files in /var/lib/asterisk/sounds/custom and do with them what you want. on Friday 09/15/2006 Nick Ellson([EMAIL PROTECTED]) wrote I seem to have stumped myself on this one. I had my son rattle off some really great sound bytes for his own extension (busy, after hours, etc) and that was easy to set up with the dial plan. Now I have his actual VM greeting in a .gsm and no idea how to get it into his VM Greeting, I am guessing that these are not stored where the other sounds are, maybe in the database? I looked through the .pdf book, not many helpful hits on google. Help? :) Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] saved.gsm - Voicemail greeting ??
I assume it will use the files .gsm too? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Friday, September 15, 2006 10:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] saved.gsm - Voicemail greeting ?? Example for mailbox 100 under context default /var/spool/asterisk/voicemail/default/100 -rwx-w 1 asterisk asterisk 442604 Aug 22 16:44 busy.wav -rwx-w 1 asterisk asterisk 44976 Aug 22 16:44 busy.WAV -rwx-w 1 asterisk asterisk 25964 Aug 8 02:17 greet.wav -rwx-w 1 asterisk asterisk 2660 Aug 8 02:17 greet.WAV drwx-w 2 asterisk asterisk 4096 Sep 13 13:45 INBOX drwx-w 2 asterisk asterisk 4096 Sep 13 13:37 tmp -rwx-w 1 asterisk asterisk 168044 Aug 9 17:00 unavail.wav -rwx-w 1 asterisk asterisk 17090 Aug 9 17:00 unavail.WAV drwx-w 2 asterisk asterisk 4096 Sep 1 11:31 Work Those are the files (wav format) that it expects for the voicemail greetings/name announcement. Greet.wav is the name. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Ellson Sent: Friday, September 15, 2006 10:26 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] saved.gsm - Voicemail greeting ?? Hi John, Yes, I followed an example that put all my family sound files in /var/lib/asterisk/sounds/local, which is also where this file is. Now I am trying to figure out how to get the unavailable|name|Busy .gsm's I made loaded into a mailbox without playing my sounds back into a phone ;) Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Fri, 15 Sep 2006, John covici wrote: Check in /var/spool/asterisk/voicemail/default/extension number for a particular extension, don't know how you want to differentiate after hours, etc. Also, you can put files in /var/lib/asterisk/sounds/custom and do with them what you want. on Friday 09/15/2006 Nick Ellson([EMAIL PROTECTED]) wrote I seem to have stumped myself on this one. I had my son rattle off some really great sound bytes for his own extension (busy, after hours, etc) and that was easy to set up with the dial plan. Now I have his actual VM greeting in a .gsm and no idea how to get it into his VM Greeting, I am guessing that these are not stored where the other sounds are, maybe in the database? I looked through the .pdf book, not many helpful hits on google. Help? :) Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] saved.gsm - Voicemail greeting ??
Part of the directory as well as when you get to leave that person a VM (instead of saying the user at extension blah blah blah is unavailable it will read back the greeting file) are the 2 places I have heard it so far. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Ellson Sent: Friday, September 15, 2006 11:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] saved.gsm - Voicemail greeting ?? Trying that now... umm, anyone know what condition makes use of just the name in voicemail, is that part of the directory or something? -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Fri, 15 Sep 2006, Bill Gibbs wrote: I assume it will use the files .gsm too? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Friday, September 15, 2006 10:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] saved.gsm - Voicemail greeting ?? Example for mailbox 100 under context default /var/spool/asterisk/voicemail/default/100 -rwx-w 1 asterisk asterisk 442604 Aug 22 16:44 busy.wav -rwx-w 1 asterisk asterisk 44976 Aug 22 16:44 busy.WAV -rwx-w 1 asterisk asterisk 25964 Aug 8 02:17 greet.wav -rwx-w 1 asterisk asterisk 2660 Aug 8 02:17 greet.WAV drwx-w 2 asterisk asterisk 4096 Sep 13 13:45 INBOX drwx-w 2 asterisk asterisk 4096 Sep 13 13:37 tmp -rwx-w 1 asterisk asterisk 168044 Aug 9 17:00 unavail.wav -rwx-w 1 asterisk asterisk 17090 Aug 9 17:00 unavail.WAV drwx-w 2 asterisk asterisk 4096 Sep 1 11:31 Work Those are the files (wav format) that it expects for the voicemail greetings/name announcement. Greet.wav is the name. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Ellson Sent: Friday, September 15, 2006 10:26 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] saved.gsm - Voicemail greeting ?? Hi John, Yes, I followed an example that put all my family sound files in /var/lib/asterisk/sounds/local, which is also where this file is. Now I am trying to figure out how to get the unavailable|name|Busy .gsm's I made loaded into a mailbox without playing my sounds back into a phone ;) Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Fri, 15 Sep 2006, John covici wrote: Check in /var/spool/asterisk/voicemail/default/extension number for a particular extension, don't know how you want to differentiate after hours, etc. Also, you can put files in /var/lib/asterisk/sounds/custom and do with them what you want. on Friday 09/15/2006 Nick Ellson([EMAIL PROTECTED]) wrote I seem to have stumped myself on this one. I had my son rattle off some really great sound bytes for his own extension (busy, after hours, etc) and that was easy to set up with the dial plan. Now I have his actual VM greeting in a .gsm and no idea how to get it into his VM Greeting, I am guessing that these are not stored where the other sounds are, maybe in the database? I looked through the .pdf book, not many helpful hits on google. Help? :) Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list
[asterisk-users] Third Lane PBX Manger Multi-Tenant
Anyone using that product with success? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] QuadBRI and Zyxel Wifi phone stop working togetherafter 3 calls
Make those calls then check the CLI sip show channels and see if the channels are stay up -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Frederik Fix Sent: Wednesday, September 13, 2006 8:35 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] QuadBRI and Zyxel Wifi phone stop working togetherafter 3 calls Hi, I have a strange problem that I have no idea how to debug: I have a Zyxel Prestige 2000W Wifi telephone that is connected to my Asterisk server which has a Junghanns.net QuadBRI card. I can make exactly 3 calls to the outside over the QuadBRI. Any calls after that fail with the log saying that all lines are busy. Turning the phone off and on solves the problem and I can make 3 calls again before it repeats. This problem does not occur when I make calls from my Cisco 7960G phones using SCCP or using eyebeam and SIP. Also making calls from the Zyxel through a cheap Cologne chipset ISDN card using zaphfc does not show this problem. I am using the following versions: Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1r Zyxel Prestige 2000W (version 1) Zyxel-Firmware: Wj.00.11 Any help is very much appreciated, Frederik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Receive Fax with rxfax on asterisk with debian
I just ran into this situation 15 mins ago
and I installed NvFaxDetect and it works great so farI tested it out
with a few one page and a couple of multi page faxes and all worked.
http://www.voip-info.org/wiki-NVFaxDetect
Bill
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dominik Weber
Sent: Monday, September 11, 2006
2:48 PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
Receive Fax with rxfax on asterisk with debian
Have nobody any idea or tipps for me ?
- Original Message -
From: Dominik Weber
To: asterisk-users@lists.digium.com
Sent: Saturday,
September 09, 2006 8:34 AM
Subject: [asterisk-users]
Receive Fax with rxfax on asterisk with debian
Hello,
my name is dominik, and i'm using asterisk with voip without
isdn, only sip.
I'm using Asterisk Version 1.0.7 on Debian 3.0.
I've configured the fax receive in the
/etc/asterisk/extensions.conf:
exten =
99,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = 99,2,rxfax(${FAXFILE})
exten = 99,3,Hangup
In the Debuglevel i see, while i send a fax,that he wants to
write the tif file.
But on my sending machine i got the error 3311 the
number isn't a g3 fax.
On asterisk i don't find any errors.
When i call the number with a telefon i got the fax sound.
Can you help me ?
Gruß Dom
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cisco PIX firewall and nat=yes
Title: Message As a follow up those commands helped with the outbound calls but inbound still had issues. Asterisk would still show the peer UNREACHABLE. Turning off qualify has fixed the problem! Bill From: Bill D'Anjou [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 23, 2006 12:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Bill Gibbs Subject: RE: [asterisk-users] Cisco PIX firewall and nat=yes You might need: fixup protocol sip 5060 fixup protocol sip udp 5060 in the PIX if these commands aren't supported you might need newer code. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Wednesday, August 23, 2006 8:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco PIX firewall and nat=yes I have a Polycom 501 that works great from behind simple firewalls, like Dlink, etc however behind a Cisco PIX Firewall I see the register messages for the extensions on the Asterisk CLI but when I do a sip show peers I see: 702/702 x.x.x.x D N 54297 UNREACHABLE 701/701 x.x.x.x D N 54297 UNREACHABLE 700/700 x.x.x.x D N 54297 UNREACHABLE But I see stuff like n Registered SIP '702' at x.x.x.x port 54297 expires 60 I have a single phone with multiple extensions in the example above. As a test I changed that phone to a single extension (700), I see the Registered line but it still says UNREACHABLE. I know the Asterisk config is good because every device (soft, hard phone) works and I know the NAT works because Ive tested that out. SoIm thinking it has something to do with the PIX. Any ideas? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cisco PIX firewall and nat=yes
Thanks I will check into this. I don't actually have access to the PIX (I have to talk to like 3 people to get to the person who actually manages this for the client) ...but that makes sense too I currently have it registering at 60 secs -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peder @ NetworkOblivion Sent: Wednesday, September 06, 2006 12:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco PIX firewall and nat=yes There is a Timeout SIP in the config. What is it set to? If it is less than the the qualify interval, which I believe is 60 seconds, then the PIX will close the inbound hole for qualify traffic. We've got lots of phones at several remote sites all running behind PIX's and all being NAT'd to the same IP (per location) and everything works perfect if qualify is on. If we disable qualify, then the SIP inbound hole gets closed per the Timeout SIP and calls don't go through until the phone re-registers and the hole opens again (they can still call out). Bill Gibbs wrote: As a follow up those commands helped with the outbound calls but inbound still had issues. Asterisk would still show the peer UNREACHABLE. Turning off qualify has fixed the problem! Bill *From:* Bill D'Anjou [mailto:[EMAIL PROTECTED] *Sent:* Wednesday, August 23, 2006 12:47 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Cc:* Bill Gibbs *Subject:* RE: [asterisk-users] Cisco PIX firewall and nat=yes You might need: fixup protocol sip 5060 fixup protocol sip udp 5060 in the PIX if these commands aren't supported you might need newer code. Bill -Original Message- *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Bill Gibbs *Sent:* Wednesday, August 23, 2006 8:53 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Cisco PIX firewall and nat=yes I have a Polycom 501 that works great from behind simple firewalls, like Dlink, etc however behind a Cisco PIX Firewall I see the register messages for the extensions on the Asterisk CLI but when I do a sip show peers I see: 702/702x.x.x.x D N 54297 UNREACHABLE 701/701x.x.x.x D N 54297 UNREACHABLE 700/700x.x.x.x D N 54297 UNREACHABLE But I see stuff like n Registered SIP '702' at x.x.x.x port 54297 expires 60 I have a single phone with multiple extensions in the example above. As a test I changed that phone to a single extension (700), I see the Registered line but it still says UNREACHABLE. I know the Asterisk config is good because every device (soft, hard phone) works and I know the NAT works because I've tested that out. So...I'm thinking it has something to do with the PIX. Any ideas? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Network stuff you didn't know http://www.networkoblivion.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call parking with Polycom's - works but MOH stops in one scenario
501s, 601s running 1.6.5 Asterisk 1.2.10 NAT Logs at the bottom of the email Using AMP or FreePBX for the config files Heres whats happening: Call comes in Answer the call On the Polycom Hit Transfer (person calling in hears MOH just fine) Enter park extension (my case 190) Listen to the digits being read back, ie 191 Now heres where it gets odd 1) If I wait after the digits are read back MOH starts on the Polycom At this point if I hit Transfer (complete the attended transfer obviously) to send the call to the park, MOH stops on the phone calling in. However parking continues to work, including ring backso other than the MOH stopping (which I assume has to do something with Asterisk not thinking the phone is on hold anymore?) the park feature works fine. Basically if I wait too long it never initiates the MOH again to the parked call however - is this because Asterisk thinks its now on a real extension hence no MOH (which may be why it plays back MOH if I transfer the call to the park extension in the first place) The call DOES show up in show parkedcalls though so Asterisk obviously knows it is parked. 2) Now if I hit Transfer quickly after the digits are read back (basically before it finishes the last digit but enough for me to know what it is), MOH continues on the person who is parked (which is the behavior I want) Park pickup/ringback works fine.and the logs show this 3) Blind transfer MOH always works but not what I want since the person who put it on Park now doesnt know what extension to pick it up I also see in the logs after the final transfer -- Incoming call: Got SIP response 500 Internal Server Error back from 192.168.1.100 No matter if I hit Transfer the 2nd time quickly or wait for MOH to start On a side note, anyone ever get the Polycom call-park and Park softkey feature to work? There doesnt seem to be any documentation about it and hitting the button does nothing. Music on hold plays (transfer quick, just before the last digit is read back) Notice the MOH starts for the parked call after the ZOMBIE lines -- SIP/102-09515c68 is ringing -- SIP/102-09515c68 answered SIP/X.X.X.X-094fc5d8 -- Started music on hold, class 'default', on SIP/X.X.X.X-094fc5d8 -- Executing Park(SIP/102-09590970, ) in new stack == Parked SIP/102-09590970 on 191. Will timeout back to extension [from-internal] s, 1 in 45 seconds -- Added extension '191' priority 1 to parkedcalls -- Playing 'digits/1' (language 'en') -- Playing 'digits/9' (language 'en') -- Playing 'digits/1' (language 'en') -- Stopped music on hold on SIP/X.X.X.X-094fc5d8 == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/102-09590970ZOMBIE' in macro 'dial' == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/102-09590970ZOMBIE' in macro 'exten-vm' == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/102-09590970ZOMBIE' -- Started music on hold, class 'default', on SIP/X.X.X.X-094fc5d8 == Spawn extension (from-internal, s, 1) exited KEEPALIVE on 'SIP/X.X.X.X-094fc5d8' -- Incoming call: Got SIP response 500 Internal Server Error back from 192.168.1.100 so I see how it started, then stopped, then started MOH again now if I do it after I wait after the last digit is read and then hit Transfer here's the log output: It never initiates the MOH is this because Asterisk thinks its now on a real extension hence no MOH Notice how the started MOH is initiated BEFORE the lines about the ZOMBIE stuff than above where it worked -- SIP/102-094f9970 is ringing -- SIP/102-094f9970 answered SIP/X.X.X.X-09515c68 -- Started music on hold, class 'default', on SIP/X.X.X.X-09515c68 -- Executing Park(SIP/102-0950a2b0, ) in new stack == Parked SIP/102-0950a2b0 on 191. Will timeout back to extension [from-internal] s, 1 in 45 seconds -- Added extension '191' priority 1 to parkedcalls -- Playing 'digits/1' (language 'en') -- Playing 'digits/9' (language 'en') -- Playing 'digits/1' (language 'en') -- Started music on hold, class 'default', on SIP/102-0950a2b0 == Spawn extension (from-internal, s, 1) exited KEEPALIVE on 'SIP/102-0950a2b0' -- Stopped music on hold on SIP/102-0950a2b0 -- Stopped music on hold on SIP/X.X.X.X-09515c68 -- Started music on hold, class 'default', on SIP/X.X.X.X-09515c68 == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/102-0950a2b0ZOMBIE' in macro 'dial' == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/102-0950a2b0ZOMBIE' in macro 'exten-vm' == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/102-0950a2b0ZOMBIE' -- Incoming call: Got SIP response 500 Internal Server Error back from 192.168.1.100 Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Call parking with Polycom's - works but MOH stops in one scenario
When using the # key identified in features.conf this issue goes away. Stilloddand so is the lack of documentation on the built in Park button Bill From: Bill Gibbs Sent: Monday, August 28, 2006 1:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Call parking with Polycom's - works but MOH stops in one scenario 501s, 601s running 1.6.5 Asterisk 1.2.10 NAT Logs at the bottom of the email Using AMP or FreePBX for the config files Heres whats happening: Call comes in Answer the call On the Polycom Hit Transfer (person calling in hears MOH just fine) Enter park extension (my case 190) Listen to the digits being read back, ie 191 Now heres where it gets odd 1) If I wait after the digits are read back MOH starts on the Polycom At this point if I hit Transfer (complete the attended transfer obviously) to send the call to the park, MOH stops on the phone calling in. However parking continues to work, including ring backso other than the MOH stopping (which I assume has to do something with Asterisk not thinking the phone is on hold anymore?) the park feature works fine. Basically if I wait too long it never initiates the MOH again to the parked call however - is this because Asterisk thinks its now on a real extension hence no MOH (which may be why it plays back MOH if I transfer the call to the park extension in the first place) The call DOES show up in show parkedcalls though so Asterisk obviously knows it is parked. 2) Now if I hit Transfer quickly after the digits are read back (basically before it finishes the last digit but enough for me to know what it is), MOH continues on the person who is parked (which is the behavior I want) Park pickup/ringback works fine.and the logs show this 3) Blind transfer MOH always works but not what I want since the person who put it on Park now doesnt know what extension to pick it up I also see in the logs after the final transfer -- Incoming call: Got SIP response 500 Internal Server Error back from 192.168.1.100 No matter if I hit Transfer the 2nd time quickly or wait for MOH to start On a side note, anyone ever get the Polycom call-park and Park softkey feature to work? There doesnt seem to be any documentation about it and hitting the button does nothing. Music on hold plays (transfer quick, just before the last digit is read back) Notice the MOH starts for the parked call after the ZOMBIE lines -- SIP/102-09515c68 is ringing -- SIP/102-09515c68 answered SIP/X.X.X.X-094fc5d8 -- Started music on hold, class 'default', on SIP/X.X.X.X-094fc5d8 -- Executing Park(SIP/102-09590970, ) in new stack == Parked SIP/102-09590970 on 191. Will timeout back to extension [from-internal] s, 1 in 45 seconds -- Added extension '191' priority 1 to parkedcalls -- Playing 'digits/1' (language 'en') -- Playing 'digits/9' (language 'en') -- Playing 'digits/1' (language 'en') -- Stopped music on hold on SIP/X.X.X.X-094fc5d8 == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/102-09590970ZOMBIE' in macro 'dial' == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/102-09590970ZOMBIE' in macro 'exten-vm' == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/102-09590970ZOMBIE' -- Started music on hold, class 'default', on SIP/X.X.X.X-094fc5d8 == Spawn extension (from-internal, s, 1) exited KEEPALIVE on 'SIP/X.X.X.X-094fc5d8' -- Incoming call: Got SIP response 500 Internal Server Error back from 192.168.1.100 so I see how it started, then stopped, then started MOH again now if I do it after I wait after the last digit is read and then hit Transfer here's the log output: It never initiates the MOH is this because Asterisk thinks its now on a real extension hence no MOH Notice how the started MOH is initiated BEFORE the lines about the ZOMBIE stuff than above where it worked -- SIP/102-094f9970 is ringing -- SIP/102-094f9970 answered SIP/X.X.X.X-09515c68 -- Started music on hold, class 'default', on SIP/X.X.X.X-09515c68 -- Executing Park(SIP/102-0950a2b0, ) in new stack == Parked SIP/102-0950a2b0 on 191. Will timeout back to extension [from-internal] s, 1 in 45 seconds -- Added extension '191' priority 1 to parkedcalls -- Playing 'digits/1' (language 'en') -- Playing 'digits/9' (language 'en') -- Playing 'digits/1' (language 'en') -- Started music on hold, class 'default', on SIP/102-0950a2b0 == Spawn extension (from-internal, s, 1) exited KEEPALIVE on 'SIP/102-0950a2b0' -- Stopped music on hold on SIP/102-0950a2b0 -- Stopped music on hold on SIP/X.X.X.X-09515c68 -- Started music on hold, class 'default', on SIP/X.X.X.X-09515c68 == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/102-0950a2b0ZOMBIE' in macro 'dial' == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/102-0950a2b0ZOMBIE' in macro 'exten-vm' == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP
[asterisk-users] DNS
Asterisk server is setup in /etc/resolv.conf to query my primary and backup NS. Had an issue with my primary NS and asterisk refused to complete any calls or forward inbound calls to extensions. I had to manually switch it to look at the backup NS first then reboot for it to start working while I fixed the primary. Is this behavior normal or am I missing a step? All hosts, etc are identified by IP. Ver 1.2.10 Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco PIX firewall and nat=yes
I have a Polycom 501 that works great from behind simple firewalls, like Dlink, etc however behind a Cisco PIX Firewall I see the register messages for the extensions on the Asterisk CLI but when I do a sip show peers I see: 702/702 x.x.x.x D N 54297 UNREACHABLE 701/701 x.x.x.x D N 54297 UNREACHABLE 700/700 x.x.x.x D N 54297 UNREACHABLE But I see stuff like n Registered SIP '702' at x.x.x.x port 54297 expires 60 I have a single phone with multiple extensions in the example above. As a test I changed that phone to a single extension (700), I see the Registered line but it still says UNREACHABLE. I know the Asterisk config is good because every device (soft, hard phone) works and I know the NAT works because Ive tested that out. SoIm thinking it has something to do with the PIX. Any ideas? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cisco PIX firewall and nat=yes
Also the phone can dial out from behind the PIXbut obviously not receive calls. Bill From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Wednesday, August 23, 2006 11:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco PIX firewall and nat=yes I have a Polycom 501 that works great from behind simple firewalls, like Dlink, etc however behind a Cisco PIX Firewall I see the register messages for the extensions on the Asterisk CLI but when I do a sip show peers I see: 702/702 x.x.x.x D N 54297 UNREACHABLE 701/701 x.x.x.x D N 54297 UNREACHABLE 700/700 x.x.x.x D N 54297 UNREACHABLE But I see stuff like n Registered SIP '702' at x.x.x.x port 54297 expires 60 I have a single phone with multiple extensions in the example above. As a test I changed that phone to a single extension (700), I see the Registered line but it still says UNREACHABLE. I know the Asterisk config is good because every device (soft, hard phone) works and I know the NAT works because Ive tested that out. SoIm thinking it has something to do with the PIX. Any ideas? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] PRI problems - no D channel
What does the telco say when they test the circuit? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron Gage Sent: Thursday, August 17, 2006 2:32 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PRI problems - no D channel Quoting C F [EMAIL PROTECTED]: My guess is it's some Intel mobo. Did you restart the system since? If you did that might be the problem, try restarting and unplug the power for at least 60 seconds before powering it back up. That was the first thing I tried: first trying to unload/reload the wct1xxp and zaptel modules, then reload them, then tried rebooting the computer (multiple times), then this morning, I gave it a 30 minute time out. No effect - still getting the D-chan errors. Unfortunately, the system is some 90 miles north of here so I can't verify if anything on the Adtran has changed or not (or reseat cables). My remote hands aren't available right now either so I can't verify anything regarding the circuit at this time. The fact that I am not getting any error reports at all about the transport (HDLC type errors) tends to make me think that the circuit is fine. I would have to imagine that if the channel switched from PRI to T1 it would throw all kinds of errors. Same thing if the signaling or buildout or what not was incorrect. The only errors I am seeing are the ones about the D channel not being there, and one (probably quite related) about head of queue has not been transmitted yet. Ron On 8/17/06, Ron Gage [EMAIL PROTECTED] wrote: The PRI is connected at one end to an Adtran Atlas, I believe a 600. The other end of the PRI is connected to a Digium T100. The two are seperated by roughly 10 foot of cat-5. Motherboard is whatever Dell put into their Precision 530MT line of workstations. Like I said, it worked just fine yesterday and today I am getting D-Channel errors. Thanks for your assistance! Ron Quoting C F [EMAIL PROTECTED]: What is the PRI connected to: What hardware for the T1? What Motherboard? On 8/17/06, Ron Gage [EMAIL PROTECTED] wrote: Hey guys: I am having a bit of a problem with our PRI under Asterisk. I am seeing the following error every 10 seconds... Aug 17 08:54:55 WARNING[2458]: chan_zap.c:2289 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! Of course, I have the d-chan defined in /etc/zaptel.conf... loadzone = us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 Neat thing is: this worked yesterday and isn't working today (Asterisk isn't answering the PRI on any inbound DID). zttool shows no problems with the T100 and no alarm conditions. The PRI is being drove by an Adtran Atlas. HELP! Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] PRI problems - no D channel
I know but you could save some time and have it tested while waiting...they might find a problem and save you a lot of headache. I can tell you are one of the rare people who actually checks their stuff before calling anyone but like another posted said, D Channels tend to be provider related for some reason! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron Gage Sent: Thursday, August 17, 2006 3:29 PM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] PRI problems - no D channel Quoting Bill Gibbs [EMAIL PROTECTED]: What does the telco say when they test the circuit? Bill Bill: I am having my remote hands check first on the Adtran that is feeding the Asterisk box, then then go upstream from there. Thanks for helping me see the obvious path to follow! :) Ron -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron Gage Sent: Thursday, August 17, 2006 2:32 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PRI problems - no D channel Quoting C F [EMAIL PROTECTED]: My guess is it's some Intel mobo. Did you restart the system since? If you did that might be the problem, try restarting and unplug the power for at least 60 seconds before powering it back up. That was the first thing I tried: first trying to unload/reload the wct1xxp and zaptel modules, then reload them, then tried rebooting the computer (multiple times), then this morning, I gave it a 30 minute time out. No effect - still getting the D-chan errors. Unfortunately, the system is some 90 miles north of here so I can't verify if anything on the Adtran has changed or not (or reseat cables). My remote hands aren't available right now either so I can't verify anything regarding the circuit at this time. The fact that I am not getting any error reports at all about the transport (HDLC type errors) tends to make me think that the circuit is fine. I would have to imagine that if the channel switched from PRI to T1 it would throw all kinds of errors. Same thing if the signaling or buildout or what not was incorrect. The only errors I am seeing are the ones about the D channel not being there, and one (probably quite related) about head of queue has not been transmitted yet. Ron On 8/17/06, Ron Gage [EMAIL PROTECTED] wrote: The PRI is connected at one end to an Adtran Atlas, I believe a 600. The other end of the PRI is connected to a Digium T100. The two are seperated by roughly 10 foot of cat-5. Motherboard is whatever Dell put into their Precision 530MT line of workstations. Like I said, it worked just fine yesterday and today I am getting D-Channel errors. Thanks for your assistance! Ron Quoting C F [EMAIL PROTECTED]: What is the PRI connected to: What hardware for the T1? What Motherboard? On 8/17/06, Ron Gage [EMAIL PROTECTED] wrote: Hey guys: I am having a bit of a problem with our PRI under Asterisk. I am seeing the following error every 10 seconds... Aug 17 08:54:55 WARNING[2458]: chan_zap.c:2289 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! Of course, I have the d-chan defined in /etc/zaptel.conf... loadzone = us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 Neat thing is: this worked yesterday and isn't working today (Asterisk isn't answering the PRI on any inbound DID). zttool shows no problems with the T100 and no alarm conditions. The PRI is being drove by an Adtran Atlas. HELP! Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ron Gage - Westland MI [EMAIL PROTECTED
[asterisk-users] Intel D945G chipset
Any problem running Asterisk w/ Digium hardware with motherboards using that chipset (for example the D945GPM) ftp://download.intel.com/design/motherbd/pm/D3610601US.pdf I was thinking of running a TE212P. Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Page Groups
For paging, and I have not done this yet,
you would probably have to invite all the phones to a conference with the
auto-answer
The below works great for intercom though .
Polycom which I have used
exten = _*7XXX,1,SetVar(ALERT_INFO=Ring
Answer)
exten = _*7XXX,2,Dial.blah
Bill
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Langstaff
Sent: Tuesday, August 15, 2006
12:46 PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Page
Groups
For intercom, do you mean placing a call
that is automatically answered by the called party?
If so, the following works for legacy
phones connected via a Citel Handset Gateway, amongst others:
exten = _*803X.,1,Macro(user-callerid)
exten = _*803X.,2,SetVar(_ALERT_INFO=info=alert-autoanswer)
exten =
_*803X.,3,SIPAddHeader(Answer-Mode: Auto)
exten = _*803X.,4,Dial(SIP/${EXTEN:4})
(so you dial *803 and then the extension
number you want to target)
Similar techniques can be used for page.
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Curt Shaffer
Sent: 15 August 2006 17:16
To: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Page
Groups
I have a company that I am going to be moving away from a
legacy PBX to Asterisk. They use page zones pretty heavy and I would like to
keep that functionality. Basically when someone is not at their desk the
receptionist pages all of the phones, telling them there is a call. Does anyone
out there know of the best phones to do this with and if it is really even
possible. I see that intercom is not supported and paging appears to be
minimally supported.
Thanks
Curt
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Ever donate Software to Digium? If you didyoura fool.
Not only that but Asterisk and Digium has enabled ALL of us to market and produce and support a product for businesses that would have no other alternative but to spend even more money on the big boys or get smaller less featured phone systems without the benefits of VOIP. We all succeed in this scenario and the resources Digium has put into this product has helped us just as much (if not more) than it has helped them. You only see this type of jealousy from people who haven't made an impact, open or closed source. I see people on the lists complaining about having to pay $10 for a g729 codec or that some of the digital interface cards are a lot of money - that's such a small thing to complain about when you are getting Asterisk for $0 and it's enabled you to make money! Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MBIT Technologies Sent: Wednesday, August 09, 2006 7:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Ever donate Software to Digium? If you didyoura fool. If you gave software to Digium then you helped Mark become very rich. What's wrong with making Mark a rich man? He has come up with a great new product and I'm sure he has risked a lot to get it to you. Asterisk is free so he owes you nothing. How about you take your jealousy elsewhere or maybe put your energy into doing something worthwhile. Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 4950 5609 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of George Gardiner Sent: Wednesday, 9 August 2006 8:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Ever donate Software to Digium? If you did youra fool. With respect, I don't think you understand the dynamics of growing a business. If we are all to benefit from the continued development of Asterisk then it is in our own best interests for Digium to succeed, because their success is for our benefit. Your posting is unfortunate as it disregards the considerable effort, cost and time put into Asterisk by Mark and Digium. By the way, I have no relation with Digium other than to derive a considerable benefit from open source software developed by Mark/Digium and a lot of other programmers, for which I am extremely grateful. Digium is not being given a whole load of money - the investors will want a slice of the company and the future profits. That's how VC funding works. Randall H. wrote: If you gave software to Digium then you helped Mark become very rich. http://abcnews.go.com/Technology/wireStory?id=2290152 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] polycom headset question
In my 1.6.6 software those options are only available in sip.cfg...can I copy that to the specific phone config file for per phone changes? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Evan P. Hall Sent: Tuesday, August 08, 2006 1:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] polycom headset question On 10:11, Tue 08 Aug 06, Dean Collins wrote: Does anyone know if there is a way of making the headset louder on the polycom 500's? The handset volume works fine but I just find the headset a little low even on the highest setting. You can set the gain for the headset, the handset, and the speakerphone all separately in the sip.cfg file or override per phone. There are two settings voice.gain.rx.digital.headset and voice.gain.rx.analog.headset. I can't remember which one you need to change. -Evan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] [OT] FYI: Polycom phone intermittent disconnects
I thought I was the only one!!! I actually replaced a phone acting just like you stated until I realized it was required the extra push as well... Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff Turner Sent: Thursday, August 03, 2006 12:48 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] [OT] FYI: Polycom phone intermittent disconnects Just a note for Polycom phone users, that will hopefully help someone. Ever since deploying an office full of Polycom 601 phones, some users have experienced intermittent disconnects, where voice transmit dies, or both receive and transmit dies. Absolutely nothing in the Asterisk logs. Solution: plug the socket into the handset in properly! Pushing the socket in, it make a nice 'click' and _seems_ to be in, but it's not (and is a bit wobbly). Push it further, until the plastic hook is not exposed at all, and it makes another click. Now it's in :) --Jeff ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] VOIP phone for Receptionist use
Title: RE: [asterisk-users] VOIP phone for Receptionist use Doesn't [EMAIL PROTECTED] need the DB flag for call waiting disabled? I believe it is *70 to enable call waiting and *71 to disable. Bill -Original Message- From: [EMAIL PROTECTED] on behalf of Jeff Busch Sent: Tue 8/1/2006 8:20 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] VOIP phone for Receptionist use I've searched through the newsgroup and online and haven't found an answer for my question... maybe I am looking for the wrong terms, I am not sure... I have a client that would like a phone that is like a typical receptionists phone. Requirements: - Ability for their 3 lines to light-up a button on the phone when one of them rings in. - Ability for the phone to ring when the receptionist is on one call and a second or third call is incoming. (this has been the biggest frustration up to now. When a second call comes, there is no tone that heard on the IP500. Perhaps I am missing a setting?) We are currently using: Asterisk @ Home 2.1 Polycom IP500/501 phones Is there a way to do what we need to using the IP500 phones? If so, can anyone give me instructions on how to make it work with [EMAIL PROTECTED] Thanks for your help in advance. Jeff ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] One way screech or tone
So would this be the remote end echo can freaking out or the Polycom on the caller side? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Wednesday, July 26, 2006 9:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] One way screech or tone Bill Gibbs wrote: Randomly, and this is very hard to debug because it happens so quickly on outbound calls I get a one way screech, it's a steady tone that's very loud. The remote end cannot hear it. You can hear the person talking through the tone. I can't describe it but it's bad enough you have to hang up and call back, and everything then is of course fine since it's so random I have not been able to reproduce it on demand. jbot: Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as screeching, feedback, static, or other useless terms. If users report static on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] One way screech or tone
Ok, in my case it would be my Cisco 3660 since that's what talks to the PRI. It talks sip to my Asterisk box. Thanks! Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Wednesday, July 26, 2006 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] One way screech or tone IP Phone - Asterisk - PSTN. This would be the Echo Canceler on the Asterisk/Zap - PSTN interface. Bill Gibbs wrote: So would this be the remote end echo can freaking out or the Polycom on the caller side? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Wednesday, July 26, 2006 9:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] One way screech or tone Bill Gibbs wrote: Randomly, and this is very hard to debug because it happens so quickly on outbound calls I get a one way screech, it's a steady tone that's very loud. The remote end cannot hear it. You can hear the person talking through the tone. I can't describe it but it's bad enough you have to hang up and call back, and everything then is of course fine since it's so random I have not been able to reproduce it on demand. jbot: Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as screeching, feedback, static, or other useless terms. If users report static on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One way screech or tone
Asterisk 1.2.4 Various hardware on the PBXs P4 3ghz, Celerons, etc Polycom phones running 1.6.5 or other ATA hardware Talking to Cisco 3660 for PRI access as well as a Cisco 3660 for long distance Ulaw or g729 AMP or FreePBX as the GUI to control Randomly, and this is very hard to debug because it happens so quickly on outbound calls I get a one way screech, its a steady tone thats very loud. The remote end cannot hear it. You can hear the person talking through the tone. I cant describe it but its bad enough you have to hang up and call back, and everything then is of course fine since its so random I have not been able to reproduce it on demand. I was going to upgrade to 1.2.10 tonight to test but was curious to see if anyone else had similar situations they have experienced. Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] emulating key system - pick up so and so on line1
Thanks allsounds like a good solution! Lets seecut their phone bill in half and get used to call parkingor continue to pay lots of money. No brainer really! Bill From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Collins Sent: Tuesday, July 18, 2006 4:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] emulating key system - pick up so and so on line1 Bruce, Good call on this one! Ive found that users can handle small changes if they are parallel with something theyre already comfortable doing. -MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves Sent: Tuesday, July 18, 2006 1:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] emulating key system - pick up so and so on line1 Bill, Our solution was to simply retrain the users to use call parking. The company had used a key system for more then a decade and I thought the change would be a tough one, but for the most part people have handled the change from Pickup line 1 to Pickup 71. Not an exact fit I know, but thought I would offer it since I was in your shoes and have found the transition easier then expected. On 7/18/06, Bill Gibbs [EMAIL PROTECTED] wrote: Is there anyway to use Polycom phones (601, 501s) to emulate a key system where you can have a shared lines that people can pick up instead of using transfer? This would make it easier for users used to putting a call on hold then telling another user so and so is on line 2. I know shared line appearances could do it but obviously that's not supported. Any other suggestions? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] emulating key system - pick up so and so on line 1
Is there anyway to use Polycom phones (601, 501s) to emulate a key system where you can have a shared lines that people can pick up instead of using transfer? This would make it easier for users used to putting a call on hold then telling another user so and so is on line 2. I know shared line appearances could do it but obviously thats not supported. Any other suggestions? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom compatible phone for Asterisk
I'm going to have to echo everyone else, the 301s are ok but the lack of full duplex speakerphone sucks, but they have a 430 now. I have a ton of 501s and 601s at clients and they are great. I do have some 300s (like the 301 without as much memory I guess) that did crash during a power outage and lost their configs but the x01 models have been fantastic and rock solid even during the same power outages. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of (AstATN) Sent: Wednesday, July 12, 2006 10:21 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom compatible phone for Asterisk Hi all, Can some one provide me the infor about polycom phones model that compatible and stable to work with Asterisk? I intend to purchase IP 300, and IP 501 models. Tq ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Provider UNREACHABLE
It's the internet...maybe for you the path to Teliax is kinda crappy? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barry Fawthrop Sent: Wednesday, July 12, 2006 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Provider UNREACHABLE Thanks All First off I never mentioned Teliax (but yes correctly ASSUMED they are my provider) and this is not a Teliax issue per se My issue is more the fact that I have Qualify = yes in sip.conf but repeatedly get REACHABLE and UNREACHABLE as can be seen below. even when I set Qualify = 3600 I still get this My question is more (a) how do I stop this ? (b) What is happening ? Thanks all Barry snip... Jul 11 13:10:38 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 57 Jul 11 13:10:48 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (117ms / 2000ms) Jul 11 13:18:01 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 13:24:17 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 2432 Jul 11 13:24:40 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (63ms / 2000ms) Jul 11 14:08:47 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 153 Jul 11 14:08:57 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (66ms / 2000ms) Jul 11 18:12:21 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 52 Jul 11 18:12:31 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (62ms / 2000ms) Jul 11 18:19:44 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 19:03:02 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 19:08:07 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 89 Jul 11 19:08:17 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 19:17:21 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 50 Jul 11 19:17:31 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 23:01:50 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 50 Jul 11 23:02:01 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) snip. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Server Optimization and Load Balancing
Keep us posted! You have a good real world load with some decent horsepower behind it so it will be interesting to see how your temporary changes you have planned in the next few days pan outI suspect the SOHO switches could be part of the problem. What is the load on the server? Bill From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mitch Jackson Sent: Tuesday, July 11, 2006 10:44 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Server Optimization and Load Balancing I'm hoping to get some guidance on some of our asterisk growing pains. Any help is greatly appreciated. Over the last few months, our call center has grown considerably and we're now experiencing choppy calls and dropped calls under full capacity. We have around a 60/40 outgoing to incoming ratio At the moment, we've got the following configuration: Asterisk SVN-trunk-r7230 All calls recorded to disk External mysql server for CDR + IVR operations Dual Xeon 2.8 4GB Ram (Dell) Digium TE210P 2 x PRI lines 72 Ploycom 301P SIP phones using ulaw codec We have a second identical server ready to offset some of the load, but we're not sure how to balance the sip phones and configuration files between the two servers. If we balance the sip registrations between the two servers, then there's the issues of both servers having to handle one call via IAX in some situations. What kind of experiences, problems and solutions have y'all had when adding servers to your center? Should we try to have incoming on one server and outgoing on the other? Should we have both servers capable of handling all the IVR operations, so the other server doesn't have to? Should we try to have an identical configuration between both servers and load balance? What kind of general optimizations should we look at to improve network / server performance? Is there a way to easily register each phone with all asterisk servers, and have the phone choose a random server to dial, and then be available as a SIP to each server if it needs to contact it? Is it a bad idea to register all phones with each server instead of distributing registration? -Here's some of the things we're got planned in the next few days: -Make sure we have all audio files in all codec formats to reduce the need for transcoding in IVR -Convert all music on hold from mp3 to native codec formats -Reduce database operations from within extensions.conf -Upgrade switches on each set of desks to midrange enterprise 100MB switches with gigabit uplinks, from SOHO netgear 100MB switches Thanks, /mitch /fidelity reserves ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Choppy MOH (Cisco gateway)
Actually this seems to have fixed it!! Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Sawa Sent: Sunday, July 09, 2006 10:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Choppy MOH (Cisco gateway) You will also want to add no vad to your dial-peer config to disable voice activity detection. I do not think it will resolve your issue, but worth a shot. -John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bill Gibbs Sent: Sunday, July 09, 2006 7:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Choppy MOH (Cisco gateway) I upgraded one of the boxes to 1.2.9.1 and using native MOH I still get it. I made sure to upgrade zaptel, etc as well. I do have something of interest to note... Placing the call on hold then taking it off hold and back on the music is ok (doing that once it gets choppy) of course this is not practical since the person using hold won't know if it's choppy. It then gets choppy again if you wait 15-20 secs. I have 2 ways of making outbound calls from all of the boxes, and I did the following via 1.2.9.1 and 1.2.4 1) Send the outbound call to the Cisco and send out via the PRI (sip phone ulaw to Cisco ulaw out the PRI) 2) Dial long distance to a provider using g729 (Polycom to Asterisk ulaw, Asterisk transcoding to g729 to provider) If I call from a sip phone OUT to my cell via the long distance provider I get no choppiness. I am not able to get inbound calls from the provider so I can only test one way. So I then switched talking to my Cisco via g729 (letting asterisk transcode ulaw to g729 and also g729 all the way through) and voice is fine but MOH is still choppy. So it must be something with the Cisco maybe? IOS version is Cisco IOS Software, 3600 Software (C3660-IS-M), Version 12.3(14)T6, RELEASE SOFTWARE (fc2) I have setup for the codecs: voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 incoming dial-peer: dial-peer voice 1 pots description Match all incoming calls, set DID incoming called-number .T direct-inward-dial forward-digits extra dial-peer voice 16 voip description to the asterisk server destination-pattern phone# voice-class codec 1 session protocol sipv2 session target ipv4:ip dtmf-relay sip-notify rtp-nte and outbound: dial-peer voice 1 pots description Outbound via PRI destination-pattern .T port 1/0:23 forward-digits all Could this have something to do with the Cisco suppressing the stream using silence suppression...I read somewhere that Asterisk relies on Sip packets for MOH??? There is not a bandwidth issue, the 3660 and boxes are on the same switch VLAN w/ DSCP enabled. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mike Sent: Monday, July 10, 2006 2:51 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Choppy MOH (Cisco gateway) i had a similar issue with the first branch of asterisk 1.2 and cheap phones (tip-100 from tatung) i'll suggest you to upgrade your asterisk box are you using bristuff ? try bristuff-0.3.0-PRE-1q which comes with asterisk 1.2.9.1 lemme know .mike On Sun, 2006-07-09 at 14:49 -0400, Bill Gibbs wrote: Cisco 3660 with a PRI talking SIP to various Asterisk boxes (not connected, separate PBXs) using ulaw all have issues with music on hold being choppy. Normal voice and SIP (taking a call from the PRI, placing a call or extension to extension calls) conversations are _perfect_ with no drop outs so it's not a problem with the PRI or the 3660 talking to the Asterisk boxes. If I call from my Polycom into an extension that immediately starts MusicOnHold it's perfect as well. However, calling into the box via the PRI and being placed on hold the music is choppy. Also, calling into an extension that spawns MusicOnHold immediately is choppy when it comes in via the Cisco. This happens with mpg123, madplay and I tried using the Asterisk 1.2 native mode in musiconhold.conf: [default] mode = files directory = /var/lib/asterisk/mohmp3 random = yes Same problem with all 3. Tried converting MP3s to a pcm or ulaw file, same problem (using lame and sox to do the conversions) It seems that this is common issue with no clear resolution. Machines are Pentium 4s 512MB or 1GB RAM. I would be the only call on the box, no load, etc. Using ztdummy (or without, same behavior) Asterisk ver 1.2.4 on all Normal voice, IVR, play back voicemail, etc are all 100% perfect only on MusicOnHold has this issue Polycom SIP phones or using X-Lite to test (used to make the call
RE: [asterisk-users] Choppy MOH (Cisco gateway)
And of course I just found this article http://www.cisco.com/warp/public/788/voice-qos/hissing.html#topic3 Hope this helps some other people out as well! Bill -Original Message- From: Bill Gibbs Sent: Monday, July 10, 2006 7:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Choppy MOH (Cisco gateway) Actually this seems to have fixed it!! Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Sawa Sent: Sunday, July 09, 2006 10:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Choppy MOH (Cisco gateway) You will also want to add no vad to your dial-peer config to disable voice activity detection. I do not think it will resolve your issue, but worth a shot. -John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bill Gibbs Sent: Sunday, July 09, 2006 7:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Choppy MOH (Cisco gateway) I upgraded one of the boxes to 1.2.9.1 and using native MOH I still get it. I made sure to upgrade zaptel, etc as well. I do have something of interest to note... Placing the call on hold then taking it off hold and back on the music is ok (doing that once it gets choppy) of course this is not practical since the person using hold won't know if it's choppy. It then gets choppy again if you wait 15-20 secs. I have 2 ways of making outbound calls from all of the boxes, and I did the following via 1.2.9.1 and 1.2.4 1) Send the outbound call to the Cisco and send out via the PRI (sip phone ulaw to Cisco ulaw out the PRI) 2) Dial long distance to a provider using g729 (Polycom to Asterisk ulaw, Asterisk transcoding to g729 to provider) If I call from a sip phone OUT to my cell via the long distance provider I get no choppiness. I am not able to get inbound calls from the provider so I can only test one way. So I then switched talking to my Cisco via g729 (letting asterisk transcode ulaw to g729 and also g729 all the way through) and voice is fine but MOH is still choppy. So it must be something with the Cisco maybe? IOS version is Cisco IOS Software, 3600 Software (C3660-IS-M), Version 12.3(14)T6, RELEASE SOFTWARE (fc2) I have setup for the codecs: voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 incoming dial-peer: dial-peer voice 1 pots description Match all incoming calls, set DID incoming called-number .T direct-inward-dial forward-digits extra dial-peer voice 16 voip description to the asterisk server destination-pattern phone# voice-class codec 1 session protocol sipv2 session target ipv4:ip dtmf-relay sip-notify rtp-nte and outbound: dial-peer voice 1 pots description Outbound via PRI destination-pattern .T port 1/0:23 forward-digits all Could this have something to do with the Cisco suppressing the stream using silence suppression...I read somewhere that Asterisk relies on Sip packets for MOH??? There is not a bandwidth issue, the 3660 and boxes are on the same switch VLAN w/ DSCP enabled. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mike Sent: Monday, July 10, 2006 2:51 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Choppy MOH (Cisco gateway) i had a similar issue with the first branch of asterisk 1.2 and cheap phones (tip-100 from tatung) i'll suggest you to upgrade your asterisk box are you using bristuff ? try bristuff-0.3.0-PRE-1q which comes with asterisk 1.2.9.1 lemme know .mike On Sun, 2006-07-09 at 14:49 -0400, Bill Gibbs wrote: Cisco 3660 with a PRI talking SIP to various Asterisk boxes (not connected, separate PBXs) using ulaw all have issues with music on hold being choppy. Normal voice and SIP (taking a call from the PRI, placing a call or extension to extension calls) conversations are _perfect_ with no drop outs so it's not a problem with the PRI or the 3660 talking to the Asterisk boxes. If I call from my Polycom into an extension that immediately starts MusicOnHold it's perfect as well. However, calling into the box via the PRI and being placed on hold the music is choppy. Also, calling into an extension that spawns MusicOnHold immediately is choppy when it comes in via the Cisco. This happens with mpg123, madplay and I tried using the Asterisk 1.2 native mode in musiconhold.conf: [default] mode = files directory = /var/lib/asterisk/mohmp3 random = yes Same problem with all 3. Tried converting MP3s to a pcm or ulaw file, same problem (using lame and sox to do the conversions) It seems that this is common issue with no clear resolution. Machines
RE: [asterisk-users] Choppy MOH (Cisco gateway)
Yes that is correct. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Monday, July 10, 2006 12:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Choppy MOH (Cisco gateway) On Jul 10, 2006, at 4:49 AM, Bill Gibbs wrote: And of course I just found this article http://www.cisco.com/warp/public/788/voice-qos/hissing.html#topic3 Hope this helps some other people out as well! So was the fix to reconfigure your gateway to not use VAD? Just want to be clear... Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Choppy MOH (Cisco gateway)
Cisco 3660 with a PRI talking SIP to various Asterisk boxes (not connected, separate PBXs) using ulaw all have issues with music on hold being choppy. Normal voice and SIP (taking a call from the PRI, placing a call or extension to extension calls) conversations are _perfect_ with no drop outs so its not a problem with the PRI or the 3660 talking to the Asterisk boxes. If I call from my Polycom into an extension that immediately starts MusicOnHold its perfect as well. However, calling into the box via the PRI and being placed on hold the music is choppy. Also, calling into an extension that spawns MusicOnHold immediately is choppy when it comes in via the Cisco. This happens with mpg123, madplay and I tried using the Asterisk 1.2 native mode in musiconhold.conf: [default] mode = files directory = /var/lib/asterisk/mohmp3 random = yes Same problem with all 3. Tried converting MP3s to a pcm or ulaw file, same problem (using lame and sox to do the conversions) It seems that this is common issue with no clear resolution. Machines are Pentium 4s 512MB or 1GB RAM. I would be the only call on the box, no load, etc. Using ztdummy (or without, same behavior) Asterisk ver 1.2.4 on all Normal voice, IVR, play back voicemail, etc are all 100% perfect only on MusicOnHold has this issue Polycom SIP phones or using X-Lite to test (used to make the call into MusicOnHold or answer the call coming in via the PRI and placing on hold) Calling in from landline or cell phone no difference Any ideas? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Choppy MOH (Cisco gateway)
I upgraded one of the boxes to 1.2.9.1 and using native MOH I still get it. I made sure to upgrade zaptel, etc as well. I do have something of interest to note... Placing the call on hold then taking it off hold and back on the music is ok (doing that once it gets choppy) of course this is not practical since the person using hold won't know if it's choppy. It then gets choppy again if you wait 15-20 secs. I have 2 ways of making outbound calls from all of the boxes, and I did the following via 1.2.9.1 and 1.2.4 1) Send the outbound call to the Cisco and send out via the PRI (sip phone ulaw to Cisco ulaw out the PRI) 2) Dial long distance to a provider using g729 (Polycom to Asterisk ulaw, Asterisk transcoding to g729 to provider) If I call from a sip phone OUT to my cell via the long distance provider I get no choppiness. I am not able to get inbound calls from the provider so I can only test one way. So I then switched talking to my Cisco via g729 (letting asterisk transcode ulaw to g729 and also g729 all the way through) and voice is fine but MOH is still choppy. So it must be something with the Cisco maybe? IOS version is Cisco IOS Software, 3600 Software (C3660-IS-M), Version 12.3(14)T6, RELEASE SOFTWARE (fc2) I have setup for the codecs: voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 incoming dial-peer: dial-peer voice 1 pots description Match all incoming calls, set DID incoming called-number .T direct-inward-dial forward-digits extra dial-peer voice 16 voip description to the asterisk server destination-pattern phone# voice-class codec 1 session protocol sipv2 session target ipv4:ip dtmf-relay sip-notify rtp-nte and outbound: dial-peer voice 1 pots description Outbound via PRI destination-pattern .T port 1/0:23 forward-digits all Could this have something to do with the Cisco suppressing the stream using silence suppression...I read somewhere that Asterisk relies on Sip packets for MOH??? There is not a bandwidth issue, the 3660 and boxes are on the same switch VLAN w/ DSCP enabled. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mike Sent: Monday, July 10, 2006 2:51 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Choppy MOH (Cisco gateway) i had a similar issue with the first branch of asterisk 1.2 and cheap phones (tip-100 from tatung) i'll suggest you to upgrade your asterisk box are you using bristuff ? try bristuff-0.3.0-PRE-1q which comes with asterisk 1.2.9.1 lemme know .mike On Sun, 2006-07-09 at 14:49 -0400, Bill Gibbs wrote: Cisco 3660 with a PRI talking SIP to various Asterisk boxes (not connected, separate PBXs) using ulaw all have issues with music on hold being choppy. Normal voice and SIP (taking a call from the PRI, placing a call or extension to extension calls) conversations are _perfect_ with no drop outs so it's not a problem with the PRI or the 3660 talking to the Asterisk boxes. If I call from my Polycom into an extension that immediately starts MusicOnHold it's perfect as well. However, calling into the box via the PRI and being placed on hold the music is choppy. Also, calling into an extension that spawns MusicOnHold immediately is choppy when it comes in via the Cisco. This happens with mpg123, madplay and I tried using the Asterisk 1.2 native mode in musiconhold.conf: [default] mode = files directory = /var/lib/asterisk/mohmp3 random = yes Same problem with all 3. Tried converting MP3s to a pcm or ulaw file, same problem (using lame and sox to do the conversions) It seems that this is common issue with no clear resolution. Machines are Pentium 4s 512MB or 1GB RAM. I would be the only call on the box, no load, etc. Using ztdummy (or without, same behavior) Asterisk ver 1.2.4 on all Normal voice, IVR, play back voicemail, etc are all 100% perfect only on MusicOnHold has this issue Polycom SIP phones or using X-Lite to test (used to make the call into MusicOnHold or answer the call coming in via the PRI and placing on hold) Calling in from landline or cell phone - no difference Any ideas? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Tadiran Coral IP PBX to Asterisk
Goal to get the CoralIP PBX long distance savings by sending it to Asterisk (which then talks via SIP to other long distance voip providers) The Coral IP supports MGCP and so does Asterisk. Has anyone tried sending calls from the Coral PBX to Asterisk via MGCP? I will be playing around with that this weekend but thought Id ask. The other way I was thinking was doing a back to back PRI, utilizing a Digium TE110P. If I understand that process correctly, using back to back PRI cards (one in the Tadiran and one in the Asterisk server) we can basically open a digital trunk to send (and accept) the calls. Any suggestions on integrating a non SIP (but VOIP) style PBX to Asterisk other than what I outlined above? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Tadiran Coral IP PBX to Asterisk
You are correct, I did ask the owner of the Coral to find out if it can act as a client as well, which would be perfect and save the hassle of going TDM. Do I need to be considered about QSIG when doing PRI crossover? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julio Arruda Sent: Thursday, July 06, 2006 7:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Tadiran Coral IP PBX to Asterisk Bill Gibbs wrote: Goal - to get the CoralIP PBX long distance savings by sending it to Asterisk (which then talks via SIP to other long distance voip providers) The Coral IP supports MGCP and so does Asterisk. Has anyone tried sending calls from the Coral PBX to Asterisk via MGCP? I will be playing around with that this weekend but thought I'd ask. The other way I was thinking was doing a back to back PRI, utilizing a Digium TE110P. If I understand that process correctly, using back to back PRI cards (one in the Tadiran and one in the Asterisk server) we can basically open a digital trunk to send (and accept) the calls. Any suggestions on integrating a non SIP (but VOIP) style PBX to Asterisk other than what I outlined above? From what I understand, you can connect MGCP clients to Asterisk, so, if your Coral IP PBX is a mgcp client, it may work, otherwise... (quite frankly, I don't think it is the case, from wa quick search in google, it seems the Coral IP PBX is a Call Agent itself, not a media gateway/MGCP client) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voice calls sent to fax extension
Maybe their fat jowls hit a few buttons on the keypad and sent the fax tone down the line and they didn't realize it? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul A. Pringle Sent: Friday, June 23, 2006 2:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voice calls sent to fax extension I have a situation that has repeated itself a few times. Someone calls into Asterisk and is connected with a voice extension. At some point during the call, the log shows chan_zap.c: DTMF digit: f on Zap/2-1. At this point, the call is redirected to receive a fax and the Asterisk voice extension is hung up. The users report that there were no noticable tones heard just before the cutover, so I'm not sure what's going on. Is there a way to disable detection of faxes after the voicecall is initiated? We're running a Digium card to convert our analog trunks if that makes any difference. Thanks! Paul ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom Intercom - almost there
This worked great.
I made an extension 1 in context intercom and set my custom Goto
statement there (first added the SIPHeader mod)
[intercom]
exten = _XXX,1,SIPAddHeader(Alert-Info: Auto Answer)
exten = _XXX,2,Goto(from-internal,7${EXTEN},1)
and added that extension (and had it not register) and it worked fine!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kristian
Kielhofner
Sent: Wednesday, June 21, 2006 4:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom Intercom - almost there
Bill Gibbs wrote:
Ok so I added to my Freepbx config running Asterisk 1.2.4 in
extensions_custom.conf
; intercom
exten = _7XXX,1,SIPAddHeader(Alert-Info: Auto Answer)
exten = _7XXX,2,Dial(SIP/${EXTEN:1},12,Tt)
and configured my Polycoms via this page
http://www.voip-info.org/wiki/view/Polycom+auto-answer+config for auto
answer and that works fine if I dial 7 then the 3 digit extension.
No problems, the receiving end answers automatically and everything
works great.
However, since we are migrating from a key system I would like to have
one of the hard buttons, say the 3^rd line key in a 501 for example to
automatically initiate the 7 so the user only has to enter in the
extension and it automatically does the intercom style feature, thus
saving a key press.
I can easily set the 3^rd line key to generate dial tone:
In sip.conf
keys key.scrolling.timeout=1
key.IP_500.33.function.prim=Handsfree/
That makes a dialtone and I can dial as normal however I would like to
put the 7 in there automatically so they just have to dial. The
subPoint.prim function is an integer that references an array value so
that won't work.
Any ideas or suggestions? Just trying to keep the number of button
presses to a minimum.
Bill
Bill,
How about creating a special SIP user for line 3:
sip.conf:
[int]
username: int
context: int
blah blah blah
It doesn't have to (and shouldn't) register.
extensions.conf:
[int]
exten - _XXX,1,Goto(phones,7${EXTEN},1) ; make sure this context is ;
;right.
--
Kristian Kielhofner
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom Intercom - almost there
Ok so I added to my Freepbx config running Asterisk 1.2.4 in
extensions_custom.conf
; intercom
exten = _7XXX,1,SIPAddHeader(Alert-Info: Auto Answer)
exten = _7XXX,2,Dial(SIP/${EXTEN:1},12,Tt)
and configured my Polycoms via this page http://www.voip-info.org/wiki/view/Polycom+auto-answer+config
for auto answer and that works fine if I dial 7 then the 3 digit extension.
No problems, the receiving end answers automatically and
everything works great.
However, since we are migrating from a key system I would
like to have one of the hard buttons, say the 3rd line key in a 501
for example to automatically initiate the 7 so the user only has
to enter in the extension and it automatically does the intercom style feature,
thus saving a key press.
I can easily set the 3rd line key to generate
dial tone:
In sip.conf
keys key.scrolling.timeout=1
key.IP_500.33.function.prim=Handsfree/
That makes a dialtone and I can dial as normal however I
would like to put the 7 in there automatically so they just have to dial.
The subPoint.prim function is an integer that references an array value so that
wont work.
Any ideas or suggestions? Just trying to keep the
number of button presses to a minimum.
Bill
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom 601 problems with multiple registrations
This does work. I have a few phones with 1.5.something doing this. I didnt fill out the reg.x.server.x.address field so it uses the sip.cfg default. Heres a snippet of what worked on a 601 6 line keys a few days ago: reg.1.displayName=x110 reg.1.address=110 reg.1.label=x110 reg.1.type=private reg.1.thirdPartyName= reg.1.auth.userId=110 reg.1.auth.password=DURRR reg.1.server.1.address= reg.1.server.1.port= reg.1.server.1.transport=DNSnaptr reg.1.server.2.transport=DNSnaptr reg.1.server.1.expires=120 reg.1.server.1.register=1 reg.1.server.1.retryTimeOut= reg.1.server.1.retryMaxCount= reg.1.server.1.expires.lineSeize= reg.1.acd-login-logout=0 reg.1.acd-agent-available=0 reg.1.ringType=2 reg.1.lineKeys=6 reg.1.callsPerLineKey=1 If you want multiple registrations, just change the 110 and password to whatever the other extension is. Does your asterisk console show the registration? Bill From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Vincent (C) Sent: Wednesday, June 21, 2006 1:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Polycom 601 problems with multiple registrations Im stumped on this one and any help would be greatly appreciated. Im just trying to get my Polycom 601 to have multiple extensions on it. For example, on line 1 I want extension 21, on line 2 I want extension 22, and on line 3 I want extension 23. Ideally Id actually have each extension appear on 2 lines and therefore filling up all 6. I should be able to do that with the reg.x.lineKeys parameter. Anyway, Im not even at the point of getting multiple registrations to work, so Ill worry about that later. Right now the only thing that works is registering the first extension it registers just fine and works as expected. No matter what extension I put on there it works, but I only have line 1 working. What am I doing wrong? Okay, now my config. Ive got a REALLY basic set up. I copied the files off the wiki from krisk.org. I completely removed ipmid.cfg temporarily so it wouldnt interfere with this (putting it back in place has no effect). That leaves me with just sip.cfg and the phone cfg file. Im booting with FTP. I know the config files are loading correctly because I can make changes and they do have an effect. Heres the phone20.cfg file for the phone: ?xml version=1.0 encoding=UTF-8 standalone=yes? !-- Example Per-phone Configuration File -- !-- $Revision: 1.59 $ $Date: 2004/05/22 00:50:41 $ -- phone1 reg reg.1.address=21 reg.1.auth.userId=21 reg.1.auth.password=21 reg.1.server.1.address=10.20.0.1 reg.2.address=22 reg.2.auth.userId=22 reg.2.auth.password=22 reg.2.server.1.address=10.20.0.1 reg.3.address=23 reg.3.auth.userId=23 reg.3.auth.password=23 reg.3.server.1.address=10.20.0.1 / /phone1 And sip.cfg: !-- IP Application Configuration File -- !-- $Revision: 1.57.2.1 $ $Date: 2004/07/27 00:23:34 $ -- sip voIpProt local voIpProt.local.port=5060/ server voIpProt.server.1.address=10.20.0.1 voIpProt.server.1.port=5060 voIp Prot.server.1.transport=UDPonly voIpProt.server.1.expires=3600 voIpProt.serv er.1.register=1 voIpProt.server.1.retryTimeOut=0 voIpProt.server.1.retryMaxC ount=0 voIpProt.server.1.expires.lineSeize=30/ SIP voIpProt.SIP.useRFC2543hold=1 voIpProt.SIP.lcs=0 voIpProt.SIP.sendCompactHdrs=0 voIpProt.SIP.WM50=0 voIpProt.SIP.keepalive.sessionTimers=0 voIpProt.SIP.requestURI.E164.addGlobalPrefix= outboundProxy voIpProt.SIP.outboundProxy.address= voIpProt.SIP.outboundProxy.port=5060/ alertInfo voIpProt.SIP.alertInfo.1.value=AA voIpProt.SIP.alertInfo.1.class=3 / alertInfo voIpProt.SIP.alertInfo.2.value=RA voIpProt.SIP.alertInfo.2.class=4 / requestValidation voIpProt.SIP.requestValidation.1.request= voIpProt.SIP.requestValidation.1.method= voIpProt.SIP.requestValidation.1.request.1.event= digest voIpProt.SIP.requestValidation.digest.realm=10.20.0.1/ /requestValidation specialEvent voIpProt.SIP.specialEvent.lineSeize.nonStandard=1 voIpProt.SIP.specialEvent.checkSync.alwaysReboot=0/ conference voIpProt.SIP.conference.address=/ /SIP /voIpProt dialplan dialplan.impossibleMatchHandling=2 dialplan.removeEndOfDial= 1 digitmap dialplan.digitmap=[2-9]11|0T|011xxx.T|[0-1][2-9]x|[2-9]xx xxx|[2-9]xxxT dialplan.digitmap.timeOut=3/ routing server dialplan.routing.server.1.address= dialplan.routing.server.1.port=506 0/ emergency dialplan.routing.emergency.1.value=911 dialplan.routing.emergency.1 .server.1=1/ /routing /dialplan logging level change log.level.change.sip=4 log.level.change.sip.obs=5/ /level /logging /sip --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] __
RE: [Asterisk-Users] Polycom Intercom - almost there
I see what you mean. I am having one too many beers and trying to
decipher the FreePBX config files right now :)
But I can see how this will work...so I will give it a shot!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kristian
Kielhofner
Sent: Wednesday, June 21, 2006 4:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom Intercom - almost there
Bill Gibbs wrote:
Ok so I added to my Freepbx config running Asterisk 1.2.4 in
extensions_custom.conf
; intercom
exten = _7XXX,1,SIPAddHeader(Alert-Info: Auto Answer)
exten = _7XXX,2,Dial(SIP/${EXTEN:1},12,Tt)
and configured my Polycoms via this page
http://www.voip-info.org/wiki/view/Polycom+auto-answer+config for auto
answer and that works fine if I dial 7 then the 3 digit extension.
No problems, the receiving end answers automatically and everything
works great.
However, since we are migrating from a key system I would like to have
one of the hard buttons, say the 3^rd line key in a 501 for example to
automatically initiate the 7 so the user only has to enter in the
extension and it automatically does the intercom style feature, thus
saving a key press.
I can easily set the 3^rd line key to generate dial tone:
In sip.conf
keys key.scrolling.timeout=1
key.IP_500.33.function.prim=Handsfree/
That makes a dialtone and I can dial as normal however I would like to
put the 7 in there automatically so they just have to dial. The
subPoint.prim function is an integer that references an array value so
that won't work.
Any ideas or suggestions? Just trying to keep the number of button
presses to a minimum.
Bill
Bill,
How about creating a special SIP user for line 3:
sip.conf:
[int]
username: int
context: int
blah blah blah
It doesn't have to (and shouldn't) register.
extensions.conf:
[int]
exten - _XXX,1,Goto(phones,7${EXTEN},1) ; make sure this context is ;
;right.
--
Kristian Kielhofner
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail to Email on Blackberry
I couldn't find one but I didn't look too hard. To be honest, the Blackberry is so easy to use with one hand I dropped the issue. We actually switched to Windows Mobile devices which suck compared to the Blackberry for email/ease of use but I can now one click listen to my voicemail without dialing in (and using the horrible on screen only keypad of my new phone...which is the only reason I listen to the attachment via a phone now. Bill From: [EMAIL PROTECTED] on behalf of Kerry Garrison Sent: Thu 6/8/2006 12:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Voicemail to Email on Blackberry Is there any setting in the voicemail that will send the voicemail file in a type that is recognized on a Blackberry? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.techdatapros.com http://www.techdatapros.com/ winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with simple incoming calls
What do the call logs say (/var/log/asterisk/full) When in doubt - reboot via cron once a day... Personally I am running an ITSP and have yet to experience any of the issues I read about daily so I am still trying to figure out why... I am running all TDM via Cisco 3660s and all my Asterisk boxes talk to the rest of the world via SIP. 3660 - Asterisk via ulaw and a long distance provider via SIP as well (another 3660 I am talking to) Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darin Willits Sent: Tuesday, June 06, 2006 9:15 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Problem with simple incoming calls Hi all, I must admit that I am stuck. I have a TDM400P card with two FXS and two FXO modules which I had set up and configured so that it was working beautifully. The only problem was that occasionally it would get itself into a state where outgoing calls would simply be met with a very loud static. A reboot would fix this issue and everything would work fine for a while. Recently however, incoming calls have also stopped working at all. Asterisk would seemingly pick up the line only to immediately hangup. Then the internal extension will ring twice and also hang up. In trying to trace down these issues I have gone back to a bare bones configuration. zapata.conf: [channels] usecallerid=no hidecallerid=no callwaiting=no threewaycalling=no transfer=no echocancel=no echotraining=no immediate=no usedistinctiveringdetection=no ; define channels context=internal signalling=fxo_ks channel = 1 ; define channels context=incoming signalling=fxs_ks channel = 3 --- extensions.conf: [incoming] ;exten = s, 1, Answer() ;exten = s, 2, Playback(hello-world) exten = s,1, Dial(Zap/1) Neither of these configurations will currently work properly. The commented out Playback setup answers, sais it is playing back, but you don't hear anything. The Dial(zap/1) will ring the extension but as soon as you pick it up, asterisk hangs up. The debug messages from the latter example are as follows: Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:4763 zt_write: Dropping frame since I'm still dialing on Zap/1-1... Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:4763 zt_write: Dropping frame since I'm still dialing on Zap/1-1... Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:4763 zt_write: Dropping frame since I'm still dialing on Zap/1-1... Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:4763 zt_write: Dropping frame since I'm still dialing on Zap/1-1... Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:4351 __zt_exception: Exception on 20, channel 1 Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:3539 zt_handle_event: Got event Ring/Answered(2) on channel 1 (index 0) Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:1408 zt_enable_ec: No echo cancellation requested Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:1424 zt_train_ec: No echo training requested Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:3824 zt_handle_event: channel 1 answered Jun 6 21:08:42 DEBUG[2121]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'Zap/1-1' -- Zap/1-1 answered Zap/3-1 Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:4817 zt_indicate: Requested indication -1 on channel Zap/3-1 Jun 6 21:08:42 DEBUG[2134]: channel.c:2348 set_format: Set channel Zap/3-1 to read format slin Jun 6 21:08:42 DEBUG[2134]: channel.c:2348 set_format: Set channel Zap/1-1 to write format slin Jun 6 21:08:42 DEBUG[2134]: channel.c:2348 set_format: Set channel Zap/1-1 to read format slin Jun 6 21:08:42 DEBUG[2134]: channel.c:2348 set_format: Set channel Zap/3-1 to write format slin Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:2677 zt_answer: Took Zap/3-1 off hook Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:1408 zt_enable_ec: No echo cancellation requested Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:1424 zt_train_ec: No echo training requested -- Attempting native bridge of Zap/3-1 and Zap/1-1 Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:3111 zt_bridge: master: 3, slave: 1, nothingok: 0 Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:3126 zt_bridge: Stopping tones on 3/0 talking to 1/0 Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:3138 zt_bridge: Stopping tones on 1/0 talking to 3/0 Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:2954 zt_link: Making 1 slave to master 3 at 0 Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:1228 conf_add: Added 20 to conference 9/3 Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:1228 conf_add: Added 21 to conference 9/1 Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:1374 update_conf: Updated conferencing on 3, with 0 conference users Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:1374 update_conf: Updated conferencing on 1, with 0 conference users Jun 6 21:08:42 DEBUG[2121]: devicestate.c:187 do_state_change: Changing state for Zap/1 - state 2 (In use) Jun 6 21:08:42 DEBUG[2138]: app_queue.c:471 changethread: Device 'Zap/1' changed to state '2' (In use) Jun 6 21:08:42 DEBUG[2121]: devicestate.c:187 do_state_change: Changing state for Zap/3 - state 2 (In
[Asterisk-Users] Polycom - missed calls dial back
This is not necessarily Asterisk specific but if I have Polycom 301/501 and 601s and want to dial a missed call back, how do I prepend a 9 can I do this via the polycom config? I cant find anything in the docs. Bill ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
