RE: [Asterisk-Users] Asterisk Community Participant; Katrina Refugee
Call ServiceMaster :) Depends on how much charge was left in the circuit as to what will happened. If it was saltwater, probably not. Freshwater, there might be a chance that after it dries completely that it will come back online. Won't know until you can test it. Glad you and your family is safe. I have a friend who's husband is MIA still in Gulfport. Quite a time there. -Don -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JR Richardson Sent: Saturday, September 03, 2005 10:05 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Community Participant; Katrina Refugee Hi All, My family and I are doing well. Thank you all for your prayers. We are using this as an opportunity to rebuild. I didn't think I really needed to but God knows best and we will obey. My family and I will temporarily be in Lafayette, Louisiana for a while but will probably relocate to Houston, TX in the future. We already have my Daughter registered in school here. Lafayette is my old stomping ground so I'm already at home. My Wife is having a time with directions though. She went half way to Lake Charles (wrong direction) yesterday when she was coming back home from shopping. My house, office, lab and 2 vehicles back in Chalmette, LA, St Bernard Parish are swimming with the fishes, snakes and alligators along with all my computers and Asterisk application development. 100% loss, but hey, we have our health. I have both homeowners and flood insurance so I should recoup most of my losses, it will take a while to get back on track. Insurance adjusters will not be able to enter the Parish till the water is out which could take several weeks if not a few months. I was planning on speaking at this years Astricon conference in Anaheim, CA on Embedded Asterisk Systems but have to resend the invitation at this time. As you can imagine, I have other priorities. I will miss this opportunity to collaborate and share my work with this community. My FTP server is 8 feet under Lake Ponchatrain at this time and foreseeable future. My Internet provider is not online anyway but I am committed and will get my work on-line as soon as possible. I will keep up with Asterisk development as I can and will jump back into the community when available to contribute with focus and vigor. I have bought and collected equipment since being in Telecommunications, VoIP and Internet Technologies for 15 years that are irreplaceable but I will re-build my VoIP laboratory bigger and better than ever. If anyone has any trade secrets on successfully recovering waterlogged electronic equipment, please let me know. God Bless. JR Richardson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie install problem. And I alreadysearchedeverywhere!
Op zondag 4 september 2005 00:21, schreef [EMAIL PROTECTED]:
Ok - what the heck is this?v
Dial(SIP/4001-2fea, Zap/2/mycellphonenumber})
I've never seen a } used in a dial statement...
sorry, one } to many. in extensions it reads:
exten = _97.,1,Dial(Zap/2/${EXTEN:2})
still no joy though. Even changed the timeout:
Executing Dial(SIP/4001-b455, Zap/2/456343|15) in new stack
-- Called 2/456343
-- Channel 0/2, span 1 got hangup
-- Hungup 'Zap/2-1'
== No one is available to answer at this time
--
Jeroen Baten| EMAIL : [EMAIL PROTECTED]
_ __ | web : www.i2rs.nl
| )|_)(_ | tel : +31 (0)499 477 688
_|_/_| \__) | fax : +31 (0)499 476 804
Roerlaan 36, 5691 HJ, Son, the Netherlands
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Re: [Asterisk-Users] newbie install problem. And I alreadysearchedeverywhere!
Op zondag 4 september 2005 09:07, schreef Jeroen Baten:
Op zondag 4 september 2005 00:21, schreef [EMAIL PROTECTED]:
Ok - what the heck is this?v
Dial(SIP/4001-2fea, Zap/2/mycellphonenumber})
I've never seen a } used in a dial statement...
sorry, one } to many. in extensions it reads:
exten = _97.,1,Dial(Zap/2/${EXTEN:2})
still no joy though. Even changed the timeout:
Executing Dial(SIP/4001-b455, Zap/2/456343|15) in new stack
-- Called 2/456343
-- Channel 0/2, span 1 got hangup
-- Hungup 'Zap/2-1'
== No one is available to answer at this time
Now with debug info:
*CLI bri debug span 1
Enabled debugging on span 1
*CLI
-- Executing Dial(SIP/4001-923e, Zap/g1/463665) in new stack
-- Making new call for cr 133
Protocol Discriminator: Q.931 (8) len=36
Call Ref: len= 1 (reference 5/0x5) (Originator)
Message type: SETUP (5)
[04 03 80 90 a3]
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability:
Speech (0)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode
(16)
Ext: 1 User information layer 1: A-Law (35)
[18 01 89]
Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive
Dchan: 0
ChanSel: B1 channel
]
[28 04 34 30 30 31]
Display (len= 4) [ 4001 ]
[6c 06 21 81 34 30 30 31]
Calling Number (len= 8) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user number
passed network screening (1) '4001' ]
[70 07 a1 34 36 33 36 36 35]
Called Number (len= 9) [ Ext: 1 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '463665' ]
[a1]
Sending Complete (len= 1)
-- Called g1/463665
No response to SETUP message
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated, peerstate
Overlap sending
-- Channel 0/1, span 1 got hangup
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated, peerstate
Overlap sending
-- Hungup 'Zap/1-1'
== No one is available to answer at this time
Sep 4 09:20:28 NOTICE[1119251376]: rtp.c:429 ast_rtp_read: RTP: Received
packet with bad UDP checksum
Sep 4 09:20:28 WARNING[1119251376]: pbx.c:1924 ast_pbx_run: Timeout, but no
rule 't' in context 'default'
*CLI
--
Jeroen Baten| EMAIL : [EMAIL PROTECTED]
_ __ | web : www.i2rs.nl
| )|_)(_ | tel : +31 (0)499 477 688
_|_/_| \__) | fax : +31 (0)499 476 804
Roerlaan 36, 5691 HJ, Son, the Netherlands
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RE: [Asterisk-Users] Asterisk Community Participant; Katrina Refugee
Mr Richardson, I sympathize with american people after this disaster. However If i was God I would feel remorse for all people in the world in destitution because of diseases, wars, starvation, ... God should really feel remorse . Thinks to all people in destitution in the world . Harry --- JR Richardson [EMAIL PROTECTED] a écrit : Hi All, My family and I are doing well. Thank you all for your prayers. We are using this as an opportunity to rebuild. I didn't think I really needed to but God knows best and we will obey. My family and I will temporarily be in Lafayette, Louisiana for a while but will probably relocate to Houston, TX in the future. We already have my Daughter registered in school here. Lafayette is my old stomping ground so I'm already at home. My Wife is having a time with directions though. She went half way to Lake Charles (wrong direction) yesterday when she was coming back home from shopping. My house, office, lab and 2 vehicles back in Chalmette, LA, St Bernard Parish are swimming with the fishes, snakes and alligators along with all my computers and Asterisk application development. 100% loss, but hey, we have our health. I have both homeowners and flood insurance so I should recoup most of my losses, it will take a while to get back on track. Insurance adjusters will not be able to enter the Parish till the water is out which could take several weeks if not a few months. I was planning on speaking at this years Astricon conference in Anaheim, CA on Embedded Asterisk Systems but have to resend the invitation at this time. As you can imagine, I have other priorities. I will miss this opportunity to collaborate and share my work with this community. My FTP server is 8 feet under Lake Ponchatrain at this time and foreseeable future. My Internet provider is not online anyway but I am committed and will get my work on-line as soon as possible. I will keep up with Asterisk development as I can and will jump back into the community when available to contribute with focus and vigor. I have bought and collected equipment since being in Telecommunications, VoIP and Internet Technologies for 15 years that are irreplaceable but I will re-build my VoIP laboratory bigger and better than ever. If anyone has any trade secrets on successfully recovering waterlogged electronic equipment, please let me know. God Bless. JR Richardson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie install problem. And I alreadysearchedeverywhere!
Hello Jeroen,
On 04-Sep-2005 9:22, Jeroen Baten wrote:
Op zondag 4 september 2005 09:07, schreef Jeroen Baten:
Op zondag 4 september 2005 00:21, schreef [EMAIL PROTECTED]:
Ok - what the heck is this?v
Dial(SIP/4001-2fea, Zap/2/mycellphonenumber})
I've never seen a } used in a dial statement...
sorry, one } to many. in extensions it reads:
exten = _97.,1,Dial(Zap/2/${EXTEN:2})
still no joy though. Even changed the timeout:
Executing Dial(SIP/4001-b455, Zap/2/456343|15) in new stack
-- Called 2/456343
-- Channel 0/2, span 1 got hangup
-- Hungup 'Zap/2-1'
== No one is available to answer at this time
What ISDN card do you have? It sounds like a quadbri, and that your in
the netherlands, so am I. That leaves one question is it directly
connected to a kpn line or via an other pbx/pabx?
If not, you could use this as your /etc/zaptel.conf
[EMAIL PROTECTED] etc]# cat zaptel.conf | grep -v ^#
loadzone = nl
defaultzone=nl
span=1,0,3,ccs,ami
span=2,0,3,ccs,ami
span=3,0,3,ccs,ami
span=4,0,3,ccs,ami
bchan=1,2
dchan=3
bchan=4,5
dchan=6
bchan=7,8
dchan=9
bchan=10,11
dchan=12
For zapata.conf:
[EMAIL PROTECTED] asterisk]# cat zapata.conf | grep -v ^#
[channels]
switchtype = euroisdn
pridialplan = dynamic
prilocaldialplan = unknown
nationalprefix = 0
internationalprefix = 00
musiconhold=default
echocancel = yes
echocancelwhenbridged = yes
echotraining = 100
signalling = bri_cpe_ptmp
context=pstn-inbound
group = 1
channel = 1-2
signalling = bri_net_ptmp
context=isdn-inbound
group = 2
channel = 4-5
signalling = bri_net_ptmp
context=isdn-inbound
group = 3
channel = 7-8
Where port 1 is the KPN line and port 2 through 4 are internal isdn
lines where the card is acting as the network side.
The dial statements in use here:
[pstn-outbound]
exten = _0.,1,SetCallerID(${EDN_MAIN:1})
exten = _0.,2,Dial(${ZAP_MAIN}${EXTEN:1},120)
exten = _0.,3,Macro(dial-result)
Where the variables are:
[EMAIL PROTECTED] includes]# grep -e EDN_MAIN -e ZAP_MAIN ../extensions.conf
EDN_MAIN=MY PRIMARY MSN WITH FIRST ZERO
ZAP_MAIN=Zap/g1/
This works, at least here, notice the g1 instead of 2 this allows
Asterisk to select on of the 2 channels.
Asterisk output of a successfull dial:
hn00pbx01*CLI
-- Executing SetCallerID(SIP/SJCC7960-3ed1, MY MSN WITHOUT LEADING
0) in new stack
-- Executing Dial(SIP/SJCC7960-3ed1, Zap/g1/MOBILE #|120) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/MOBILE #
-- Zap/1-1 is ringing
-- Zap/1-1 answered SIP/SJCC7960-3ed1
-- Hungup 'Zap/1-1'
== Spawn extension (int-inbound, 00646142645, 2) exited non-zero on
'SIP/SJCC7960-3ed1'
FYI: when starting to mess with outgoing and incoming MSN, in asterisk
you need the full number ie: 0207712345 but for the outgoing channel you
need it without the 0, so 207712345 otherwise KPN won't accept it.
Good luck
--
Met Vriendelijke groet/Yours Sincerely
Stijn Jonker [EMAIL PROTECTED]
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[Asterisk-Users] IPSwichBoard designers wanted
I am rewriting IPSwitchBoard at the moment. I want to make IPSwitchBoard Skinable meaning that you can design your own skins with company logo etc. You will also be able to add graphical extension buttons, and leds that will light up ex. DND, busy/free, message waiting and much more. If you are graphically minded and would like to help with making test skins using PhotoShop or other tool that produces bitmaps; then please drop me an e-mail on [EMAIL PROTECTED], and I will send you some more details. Regards Thorben http://ipsoftware.thorben.dk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: [EMAIL PROTECTED] - requesting help on oh323, ISDN BRI and iConnectHere DID
I know almost nothing linux, and don't really know that much about Asterisk (proper).. but I was 'pulled' by this subject and grabbed an [EMAIL PROTECTED] installation CD (version 1.3) and just went with it. Newbie doesn't quite describe it, I'm a banker.. this simply goes to show how easy Asterisk is becoming (I certainly hope this direction was meant to be inviting to people like me). I've been at this for a little over three weeks now. I've researched these topics but have not found satisfactory answers to the following: If there are places where an answer could be found, I'd appreciate some pointer(s). oh323 Once it (oh323) was installed, I was able to dialoutfrom an h323 device with little problem. I have not, however, been able to setup the h323 device as an extension (or use it to communicate with other h323 devices, or through it to the pstn.. I was confused because in creating an extension (or a trunk) in [EMAIL PROTECTED] there was nothing to indicate that h323 was "available". The dropdown in extension creation does not include oh323 (or some such) item. I need a clue as to how to setup an extension and how to setup a trunk (using an h323 device that is). ISDN Could someone please tell me whether the AVM Fritz card is a "low pain" solution to connecting to a BRI ? I've tried connecting an ASUS card (doesn't work with the available drivers/application) so I'm really asking in order to buy. I only have one BRI - this is at home) so would appreciate any help. iConnectHere From day one, I was able to create the trunk to dialout of iConnectHere, but despite finding several claimed correct settings for receiving its DID, I've not been able to. When I connect a Grandstream 101 to the line and power it up, it has absolutely no problem receiving calls from that DID but no matter what I try, I'm unable to receive into Asterisk (@home) .. Just so we're clear, I am receiving calls from fwd and through fwd from an ipkall number.. and I'm also recieving calls from the FXO on a sipura spa3k.. - iConnectHere don't help more than point to the wiki .. (and what I find there doesn't work). Just a note to describe my handicap with linux: I'm unable to capture a log (or perhaps it's captured and I'm just not aware where).. [EMAIL PROTECTED] seems to dialout and receive calls using Macros.. I suspect it's a clever way of managing the setup, but I'm not sure where the various portions of SIP.conf, extensions.conf, extensions_additional.conf, extensions_custom.conf or indeed oh323.conf. - are relevant. I mention this because I realize no one will be able to help my "specific" problems unless I put up the confs.. but I have not added anything related to oh323 at all (excepting working on oh323.conf which is not meaningful since I have no mention of the extensions or trunks of h323 devices in the files.. and my iConnectHere 'setup' for the DID is only one of a dozen or so tries.. (which all don't work) .. ANY help would be appreciaated with these issues. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Community Participant; Katrina Refugee
It is good to see JR's faith is not shaken by then even and that he will rebuild better than ever. I hope JR's Asterisk application development was backed up somewhere so he doesnt have to re-write it all again. I have seen drives recovered from being soaked in water and it wasnt a major drama with the right equipment. The key problem as I understand it, is the kind of water - salt water being a bad thing - but still not the end of the world if the platters can be cleaned up easily. Btw Harry... This is not a discussion list for religious issues/opinion - in the positive or the negative. Criticising, or praising God here will no doubt annoy both sides of the fence. ...Skeeve ___ Skeeve Stevens, RHCE Email: [EMAIL PROTECTED] Si vis pacem, para bellum -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of harry gaillac Sent: Sunday, 4 September 2005 7:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Community Participant; Katrina Refugee Mr Richardson, I sympathize with american people after this disaster. However If i was God I would feel remorse for all people in the world in destitution because of diseases, wars, starvation, ... God should really feel remorse . Thinks to all people in destitution in the world . Harry --- JR Richardson [EMAIL PROTECTED] a écrit : Hi All, My family and I are doing well. Thank you all for your prayers. We are using this as an opportunity to rebuild. I didn't think I really needed to but God knows best and we will obey. My family and I will temporarily be in Lafayette, Louisiana for a while but will probably relocate to Houston, TX in the future. We already have my Daughter registered in school here. Lafayette is my old stomping ground so I'm already at home. My Wife is having a time with directions though. She went half way to Lake Charles (wrong direction) yesterday when she was coming back home from shopping. My house, office, lab and 2 vehicles back in Chalmette, LA, St Bernard Parish are swimming with the fishes, snakes and alligators along with all my computers and Asterisk application development. 100% loss, but hey, we have our health. I have both homeowners and flood insurance so I should recoup most of my losses, it will take a while to get back on track. Insurance adjusters will not be able to enter the Parish till the water is out which could take several weeks if not a few months. I was planning on speaking at this years Astricon conference in Anaheim, CA on Embedded Asterisk Systems but have to resend the invitation at this time. As you can imagine, I have other priorities. I will miss this opportunity to collaborate and share my work with this community. My FTP server is 8 feet under Lake Ponchatrain at this time and foreseeable future. My Internet provider is not online anyway but I am committed and will get my work on-line as soon as possible. I will keep up with Asterisk development as I can and will jump back into the community when available to contribute with focus and vigor. I have bought and collected equipment since being in Telecommunications, VoIP and Internet Technologies for 15 years that are irreplaceable but I will re-build my VoIP laboratory bigger and better than ever. If anyone has any trade secrets on successfully recovering waterlogged electronic equipment, please let me know. God Bless. JR Richardson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users iBurst Wireless Broadband from $34.95/month www.platformnetworks.net Forward undetected SPAM to: [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Open G.729 / G.723.1 update, fixed memory leak
A new release of the open source G.729 patch has been issued. The new URL is: http://www.readytechnology.co.uk/open/ipp-codecs The memory leak in codec_g729 is now fixed. This was due to a problem in a section of code copied from the Intel example. Thanks to those who assisted in locating this bug. If you are still running the old version of the codec, your Asterisk process will run out of memory after several thousand calls (for some people, that might be every 3 months, for other people, it could be more than once a day). Therefore, updating to this latest release is highly advisable. This release combines the G.729 and G.723.1 patches into a single patch against Intel's IPP sample. They are still built as separate modules, so you don't have to install both. I've also included - command line G.729 encoder for converting your WAV files into pre-recorded G.729 files - scripts for generating a Debian package. See the documentation for details. Finally, the donations page has now been fixed. Making a donation is one way to encourage programmers to contribute commercial quality code to the open source community. Regards, Daniel smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Community Participant; Katrina Refugee
I have bought and collected equipment since being in Telecommunications, VoIP and Internet Technologies for 15 years that are irreplaceable but I will re-build my VoIP laboratory bigger and better than ever. If anyone has any trade secrets on successfully recovering waterlogged electronic equipment, please let me know. God Bless. God Bless indeed. Loosing the lab and everything in it I can certainly imagine will set you back. However, I also believe that the knowledge you acquired is your best bet to rebuild and improve on what you had. Just a though I had that may give you little encouragement. Best of luck to you and your family. Regards, Senad JR Richardson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] Asterisk Community Participant; Katrina Refugee
If anyone has any trade secrets on successfully recovering waterlogged electronic equipment, please let me know. Dear JR, I am realy sorry about all this desaster, but happy to see you alive. For waterlogged equipments, no problem until they are under the water level (out of oxygen contact). The best way after that will be to clean all of this waste and poluted equipments with clear water during a long time (no more chimical products in the clean water), and to quickly dry them. Good luck. Best Regards to you and your family. Francois BERGERET, France. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dial rule / prefix with #
HI All, Have searched a bit for this one, but cant seem to find any indication of how you might use a dial prefix with the hash / pound sign as part fo the prefix. I have a gsm pod set up on asterisk which all works fine, but i want to disable cli on the outbound calls over this particular trunk. The network that the pod is on allows blocking of sending CLI by prefixing your dialed number with #31#. I have tried putting a rule into localprefixes to use this as a prefix, but if i have the # sybols in there, the rule doesnt seem to get a match. If i leave the # symbols out and just have the 31 as a prefix, the match works, but dialing the number with just 31 prefixed ends up with a invalid phone number and no call being connected. The rules i have tried are - [trunk-2] rule1=#31#+04. [trunk-2] rule1=#31#+04. [trunk-2] rule1=_#31_#+04. and [trunk-2] rule1=31+04. none of which work as desired. Is there some way of indicating a hash / pound symbol for this use or some way of escaping it so that the rule will work? Cheers, Ben ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP - How Do I Separate incoming channels from the others on a PRI
Okay, here is the background. I have a PRI with 15 active channels on it. I originally setup all of them in group=1 and all outgoing and incoming calls used this group. The phone number that I have associated with these channels ends with 750 and that is how I direct the calls. i.e. In my extensions.conf I have: exten = 750,1,Dial(SIP/120,20) All this works fine. Now I have the need to separate out three of the channels (13-15). I am using the associated phone number ending in 767 for this purpose. I have currently changed the zapata.conf to look like this: signalling=pri_cpe switchtype=national echocancel=yes echocancelwhenbridged=yes echotraining=yes callerid=asreceived group=1 context=default channel=1-12 group=2 context=fax channel=13-15 To my understanding this is supposed to separate channels 1-12 into group 1 and channels 13-15 into group 2. If that is true, that's fine, but it doesn't help me with my current issue. This is what I would like... When a call comes into the phone number ending in 750 it uses up the 12 channels in the first group, and when a call comes into the phone number ending in 767 it uses the 3 channels in group 2 AND USES NO MORE CHANNELS, only those three. The reason for all of this is that I have faxing through asterisk working and want those last three channels used for faxing but I do not want the fax lines eating up all my channels and leaving none for voice calls. My Company has evacuated from Hurricane Katrina and any help you can provide would be greatly appreciated. Thanks in advance. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Any hardphones with SIP API?
Hi! Is there any SIP hardware phone that provides an API that can be used to control and monitor the phone by external applications? Thanks in advance! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: [EMAIL PROTECTED] - requesting help on oh323, ISDN BRI and iConnectHere DID
AbdelRahman Tarzi wrote: I know almost nothing linux, and don't really know that much about Asterisk (proper).. but I was 'pulled' by this subject and grabbed an [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] installation CD (version 1.3) and just went with it. Newbie doesn't quite describe it, I'm a banker.. this simply goes to show how easy Asterisk is becoming (I certainly hope this direction was meant to be inviting to people like me). I've been at this for a little over three weeks now. I've researched these topics but have not found satisfactory answers to the following: If there are places where an answer could be found, I'd appreciate some pointer(s). Please post to the [EMAIL PROTECTED] forum: http://sourceforge.net/forum/forum.php?forum_id=420324 and/or amportal mailing list: http://lists.sourceforge.net/lists/listinfo/amportal-users or Help forum: http://sourceforge.net/forum/forum.php?forum_id=414452 oh323 Once it (oh323) was installed, I was able to dialout from an h323 device with little problem. I have not, however, been able to setup the h323 device as an extension (or use it to communicate with other h323 devices, or through it to the pstn.. I was confused because in creating an extension (or a trunk) in [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] there was nothing to indicate that h323 was available. The dropdown in extension creation does not include oh323 (or some such) item. I need a clue as to how to setup an extension and how to setup a trunk (using an h323 device that is). The current version of AMP (1.10.008) supports the creation of Custom Trunks, which in your case could be an oh323 trunk. The next version of AMP (1.10.009, currently in second beta) will support Custom extensions, which in your case would mean that you could setup your oh323 endpoint and be able to leverage AMP's dialplan. ISDN Could someone please tell me whether the AVM Fritz card is a low pain solution to connecting to a BRI ? I've tried connecting an ASUS card (doesn't work with the available drivers/application) so I'm really asking in order to buy. I only have one BRI - this is at home) so would appreciate any help. Can't comment on the pain aspect but documentation on this configuration exists: http://www.voip-info.org/tiki-index.php?page=Asterisk+AVM+Fritz+CAPI+Driver+Install (Note: this would be configured in AMP as a Custom Trunk as well) iConnectHere From day one, I was able to create the trunk to dialout of iConnectHere, but despite finding several claimed correct settings for receiving its DID, I've not been able to. When I connect a Grandstream 101 to the line and power it up, it has absolutely no problem receiving calls from that DID but no matter what I try, I'm unable to receive into Asterisk (@home) .. Just so we're clear, I am receiving calls from fwd and through fwd from an ipkall number.. and I'm also recieving calls from the FXO on a sipura spa3k.. - iConnectHere don't help more than point to the wiki .. (and what I find there doesn't work). No comment. Never heard of this VoIP Service Provider. Just a note to describe my handicap with linux: I'm unable to capture a log (or perhaps it's captured and I'm just not aware where).. Asterisk: /var/log/asterisk/full (see /etc/asterisk/logger.conf) Linux: /var/log/messages [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] seems to dialout and receive calls using Macros.. I suspect it's a clever way of managing the setup, but I'm not sure where the various portions of SIP.conf, extensions.conf, extensions_additional.conf, extensions_custom.conf or indeed oh323.conf. - are relevant. Please search the [EMAIL PROTECTED] forum and/or amportal list forum for more info on these topics. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk SMS via IAX2?
In v1.2beta1, README.sms mentions that SMS() allows sending and receiving of text messages over the PSTN (and that it's not likely to work over a compressed link.) I think this likely answers my following question, but I wanted to ask because maybe someone has accomplished this: Is it possible to route an incoming SMS via a PSTN terminated IAX2 trunk using SMS()? Ideally, I'd like to pass off any incoming SMS to my VoIP line to Kannel for further handling. Can this be done, if so, how? Cheers, HJ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] donating VOIP gear to the relief efforts.
Hey folks, The reason I ask, is I'm on my way down to north western LA to donate/help setup gear from a WISP used to own/operate. VOIP is being mentioned quite a bit. So if you have any spare gear, please donate it. We are using this site to coordinate I'm sure there are others: http://www.fastlineisp.com/volunteer Glenn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP - How Do I Separate incoming channels from the others on a PRI
This is something that you wouldn't have any control over in most cases. Your provider is the one that is sending the lines over the whole PRI, they don't and shouldn't care about which channel it comes over, they would use some system like gGrR that you can use in Asterisk to send it to the next available channel, but not over a specific one. For your problem the easiest workaround would be to have extensions setup in extensions.conf for your fax numbers, and use the setgroup checkgroup apps to make sure that you don't use more than 3 channels at onece for the 4th channel you can use the Busy or congestion app to the PRI so that the caller gets a busy or circuit busy tone. On 9/4/05, Derrick Stensrud [EMAIL PROTECTED] wrote: Okay, here is the background. I have a PRI with 15 active channels on it. I originally setup all of them in group=1 and all outgoing and incoming calls used this group. The phone number that I have associated with these channels ends with 750 and that is how I direct the calls. i.e. In my extensions.conf I have: exten = 750,1,Dial(SIP/120,20) All this works fine. Now I have the need to separate out three of the channels (13-15). I am using the associated phone number ending in 767 for this purpose. I have currently changed the zapata.conf to look like this: signalling=pri_cpe switchtype=national echocancel=yes echocancelwhenbridged=yes echotraining=yes callerid=asreceived group=1 context=default channel=1-12 group=2 context=fax channel=13-15 To my understanding this is supposed to separate channels 1-12 into group 1 and channels 13-15 into group 2. If that is true, that's fine, but it doesn't help me with my current issue. This is what I would like... When a call comes into the phone number ending in 750 it uses up the 12 channels in the first group, and when a call comes into the phone number ending in 767 it uses the 3 channels in group 2 AND USES NO MORE CHANNELS, only those three. The reason for all of this is that I have faxing through asterisk working and want those last three channels used for faxing but I do not want the fax lines eating up all my channels and leaving none for voice calls. My Company has evacuated from Hurricane Katrina and any help you can provide would be greatly appreciated. Thanks in advance. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [Asterisk-Users] Asterisk Community Participant; Katrina Refugee
For waterlogged equipments, no problem until they are under the water level (out of oxygen contact). The best way after that will be to clean all of this waste and poluted equipments with clear water during a long time (no more chimical products in the clean water), and to quickly dry them. Just to add a couple of comments to the above based on practical experience with electronics... - some hard drives are truly sealed (air tight) while others have some form of air filter for air circulation. If the drives don't have a filter, there is a high probability that simply cleaning the drive electronics and drying is all that is needed. - cleaning almost all forms of electronic equipment with tap water is acceptable, but then follow that with a rinse of distilled water. (As others have already noted, salt water is the worst to deal with after the equipment has been exposed to air.) If you can find a good printed circuit board cleaning solution, use it as a final rinse on electronics. - clean and rinse the inside of connectors (eg, usb, rj11, rj45, cable connectors) is more important then cleaning the external surface. Same with plated through holes in circuit boards. - Hair dryers (etc) can be used to help dry equipment, but be careful with assumptions relative to water that might have penetrated some components. It may take more time to dry completely then one might expect. Don't be in a hurry to fire up the hardware. Letting the equipment sit in a low humidity environment for a week or two is by far better then being in a hurry to fire it up. - Speakers will be junk regardless of cleaning methods - Its not uncommon for commercial recovery companies to suggest keeping electronic equipment, books, papers, etc, under water until it can be handed over to them. (Air contact without proper cleaning _will_ create more problems then the water itself.) - disassembling electronic equipment and cleaning all sides of printed circuit boards, etc, should be considered mandatory. (eg, remove the motherboard and clean it, remove the drive electronics and clean it.) In most cases the commercial recovery companies don't use anything more sophisticated then the processes noted above for electronics. They just have a little more hands-on experience doing it and we're all led to believe its magic. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting wcte11xp card to use IRQ
Depending on the mother board. The following might or might not help: 1. Move the card to a different PCI slot, some motherboards will assign you a different IRQ based on the PCI slot, while other motherboards might just have one PCI slot that gets its dedicated IRQ. Changing it around will tell you. 2. If your bios allows you to dedicate an IRQ to a PCI slot try it. This will just dedicate that IRQ to your digium card. 3. If your bios allows you to reserve an IRQ to legacy ISA devices, then reserve the IRQ your eth0 and digium card are sharing to that reserved mode. This will force either or both of the devices to look for a new IRQ, and might help the problem. 4. Disable anything you don't use (Sounc Card, USB, lpt, Com ports, etc.) in the bios. This will free additional IRQs. On 9/2/05, Lee Archer [EMAIL PROTECTED] wrote: Hi, is it possible to set a wcte11xp card to use a certain IRQ? I've tried a few things but it always shares the IRQ with eth0 even though the system has 4 spare ones. I can't set it via the BIOS. Regards Lee ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Sorry to write so many consecutive messages in such a short period of time, but this problem is really bugging me as it has been going on for days. When I look in Ethereal, there are actually two calls going on -- in this particular call, Source call #4 and Source call #10318, #4 coming from the asterisk server and the other one coming from my computer to the Asterisk server. I don't know why there are two separate calls, but perhaps one of you do. Its not really two separate calls; its the transmit leg and the receive leg of the same call. Anyhow, source call #10318 seems fine, sending a new packet every 20 ms pretty much perfectly and all (although I do see now that one packet has a timestamp of 33080 and the next has one of 35060 -- is this something to be concerned about? The diff is 1980, which essentially suggests there were 99 iax packets missing between those two timestamps. (Should note that timestamps miscalculations have been an issue in the iax2 source code, but I don't remember if some of the fixes were before or after the version of code your running. That really was the basis for suggesting a code upgrade.) If 99 iax2 packets are actually missing, you _would_ have a problem with the audio quality. it doesn't seem widespread). However, call #4 seems to send every 20 ms, but then there will be a pause or something in sending, in between which there will be more packets from source call #10318 which are sent pretty much OK. Keep in mind that you're looking at a full duplex flow of packets in and out. The fact that packets are not exactly one transmit for one every one received is not as important as identifying missing packets in the form of large jumps in timestamp values. Then, the next packet for source call #4 will have a timestamp of something like 33540, exactly 200 ms after the previous packet from source call #10318. However, the next packet for SC (source call) #10318 increments 20 ms like it should. Every single packet then on (in this capture, I recorded about 1500 packets) sends perfectly. iax2.rrdropped, iax2.rrjitter, and iax2.iax.rrloss returned only 2 packets--the same two, in the middle of the 1500 packets. So, out of 1500, these are the only two that seem to have problems. Asterisk can generally handle a couple of missing 20 millisecond packets, but it can't compensate (or cover up) hugh timestamp jumps. Might consider doing another ethereal run or two and see if you can reach a conclusion that the timestamp jumps are in fact associated with the choppy audio. If you can reach that conclusion, the next step is to isolate why; is it bad code or the network that's causing the issue? I don't recall from your previous postings, but could you repeat what exact code versions are running on my computer and the remote asterisk server? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Community Participant; Katrina Refugee [OT]
[EMAIL PROTECTED] wrote: If anyone has any trade secrets on successfully recovering waterlogged electronic equipment, please let me know. If it's not electrically broken, there is a big chance to save it ... open it, and wash it using hand shower and some brushes until it is clean (the best is massage hand shower - do not worry about higher pressure) if used water is hard i.e. mineralized (mostly yes) I recommend to rinse it in demineralised/destilated water (probably u can buy it on petrol stations - it's used to refill electrolyte in lead accumulators) then left it minimum 24 hours opened on free air until u are 100% sure it's dry everywhere (some devices can be left on direct sun, but LCD's or ACCU's do not like it very much) in the rest close and test it HTH, noro PS. I use this procedure to clean PC's of dust - it's fast, comfortable and easy. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP - How Do I Separate incoming channels from the others on a PRI
On Sun, 2005-09-04 at 07:39 -0500, Derrick Stensrud wrote:
Re-sending your message every 12 hours isn't nice wait at least a
couple of days, and while you wait, try to read/test more things, so
that the second time around, you can actually demonstrate that you have
progressed somewhat
Okay, here is the background. I have a PRI with 15 active channels on
it. I originally setup all of them in group=1 and all outgoing and
incoming calls used this group. The phone number that I have associated
with these channels ends with 750 and that is how I direct the calls.
i.e. In my extensions.conf I have:
exten = 750,1,Dial(SIP/120,20)
All this works fine.
Great...
Now I have the need to separate out three of the
channels (13-15). I am using the associated phone number ending in 767
for this purpose. I have currently changed the zapata.conf to look like
this:
group=1
context=default
channel=1-12
group=2
context=fax
channel=13-15
To my understanding this is supposed to separate channels 1-12 into
group 1 and channels 13-15 into group 2. If that is true, that's fine,
but it doesn't help me with my current issue. This is what I would
like... When a call comes into the phone number ending in 750 it uses
up the 12 channels in the first group, and when a call comes into the
phone number ending in 767 it uses the 3 channels in group 2 AND USES NO
MORE CHANNELS, only those three. The reason for all of this is that I
have faxing through asterisk working and want those last three channels
used for faxing but I do not want the fax lines eating up all my
channels and leaving none for voice calls. My Company has evacuated from
Hurricane Katrina and any help you can provide would be greatly appreciated.
Thanks in advance.
The above will ensure that you never have more than 12 outgoing voice
calls, or 3 outbound fax calls, but it doesn't do anything at all for
your inbound calls...
What you want is some way to indicate to the dialplan that your outbound
calls are either voice or fax, and then do something like this:
exten = _X.,1,SetGroup(${CALLTYPE}) ; This is either VOICE or FAX
exten = _X.,2,gotoif(${CALLTYPE}=VOICE,10,20)
exten = _X.,10,CheckGroup(12) ; Make sure voice calls max 12
exten = _X.,11,Goto(30)
exten = _X.,20,CheckGroup(3) ; Make sure fax calls max 3
exten = _X.,21,Goto(30)
exten = _X.,30,Dial(Zap/g1/${EXTEN})
basically, you use the setgroup/checkgroup to ensure you aren't going
over the quota for voice/fax calls. Check the syntax/etc of all the
above functions, since I am sure a lot is wrong, but it should be more
than enough to give you the right idea.
Also, for your inbound calls:
exten = 750,1,SetGroup(VOICE)
exten = 750,2,CheckGroup(12)
exten = 750,3,Dial(SIP/120,20)
exten = 750,103,Congestion
exten = 750,104,Wait(10) ; Don't need this if you use OOB indication
exten = 750,105,Hangup
exten = 767,1,SetGroup(FAX)
exten = 767,2,CheckGroup(3)
exten = 767,3,Goto(faxes,s,1)
exten = 767,103,Congestion
exten = 767,104,Wait(10) ; Don't need this if you use OOB indication
exten = 767,105,Hangup
Hope that gets you on the right track.
Basically, the thing you got wrong is that on these sorts of lines,
there is no 'connection' between the channel and the phone number. The
call arrives on the signalling channel, and a voice channel is
dynamically/randomly assigned for the call if there is one available.
Regards,
Adam
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Re: [Asterisk-Users] newbie install problem. And I alreadysearchedeverywhere!
Op zondag 4 september 2005 11:10, schreef Stijn Jonker:
Hello Jeroen,
On 04-Sep-2005 9:22, Jeroen Baten wrote:
Op zondag 4 september 2005 09:07, schreef Jeroen Baten:
Op zondag 4 september 2005 00:21, schreef [EMAIL PROTECTED]:
Ok - what the heck is this?v
Dial(SIP/4001-2fea, Zap/2/mycellphonenumber})
I've never seen a } used in a dial statement...
sorry, one } to many. in extensions it reads:
exten = _97.,1,Dial(Zap/2/${EXTEN:2})
still no joy though. Even changed the timeout:
Executing Dial(SIP/4001-b455, Zap/2/456343|15) in new stack
-- Called 2/456343
-- Channel 0/2, span 1 got hangup
-- Hungup 'Zap/2-1'
== No one is available to answer at this time
What ISDN card do you have? It sounds like a quadbri, and that your in
the netherlands, so am I. That leaves one question is it directly
connected to a kpn line or via an other pbx/pabx?
I have a BRI 2-channel Eicon Diva on KPN and yes, look at the sig, I'm dutch.
Which part of your conf should I use and not use?
kind regards,
--
Jeroen Baten| EMAIL : [EMAIL PROTECTED]
_ __ | web : www.i2rs.nl
| )|_)(_ | tel : +31 (0)499 477 688
_|_/_| \__) | fax : +31 (0)499 476 804
Roerlaan 36, 5691 HJ, Son, the Netherlands
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Re: [Asterisk-Users] unicall deploy
Hi Guilhermo. Could you share with us your experience? What is the hardware(CPU, RAM, etc) that are you using for this server? What is your Linux distribution? How many concurrent calls do you have in the high traffic moment? Which is the unicall version that are you using? Thanks a lot! D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 3252-2977 r http://www.isolve.com.br On 04 de set de 2005, at 01:06, Guillermo Freige wrote: I´m using an unicall box with 4 E1 lines getting between 6000-15000 calls per day, and between 15-30 operators using AgentLogin, all using R2 signaling to the telco and a local PBX. I´m using the Argentina variant, and using the last version of unicall 0.0.2 and asterisk 1.0.7 Guillermo From: acriollo [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non- Commercial Discussionasterisk-users@lists.digium.com To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] unicall deploy Date: Sat, 3 Sep 2005 15:04:20 -0500 Hi every one . There are any out there that have a unicall deploy working without problem ? Can give me some tips or referenece about his config ? Regards ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_sip.c:946 __sip_xmit
Hello, I am getting lots of messages like that: Sep 4 15:31:55 WARNING[10271]: chan_sip.c:946 __sip_xmit: sip_xmit of 0x835bad8 (len 758) to xxx.xxx.xxx.xxx returned -1: Invalid argument Could somebody tell me more about that warning? I would appreciate it. I could not find anything interesting about that warning. Thank you in advance. Bartosz ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Nokia 32 Terminal
Hi, Does anyone have some experience with Nokia 32 Terminal (it is an analog GSM Gateway)? After a configuration I can make only incoming calls, I'm not able to do any outgoing. Nokia signalize an error (4 short tones), when I try to phone someone. I tried postpaid simcards as well as prepaid simcards with the same result. Does anyone try to connect this gateway to Asterisk PBX if so what were the results? All the best Andrutto -- Jedyny taki czat... http://link.interia.pl/f18b0 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dlink dph-140s/ACT P104SLD
Interestingly this phone works fine if I configure it for broadvoice directly (bypassing my local AAH). All the software sip phones seem to work fine with aah. The phone does not seem to know what to do if I leave the outbound sip proxy fields blank. And itt probably does the wrong thing if I put the address of my aah server. Neither setting results in voice traffic going to the phone. With the soft sip phones, leaving the outbound proxy fields blank is apparently the way to go with a local aah server. Any ideas how to solve this? I'm still a learning but I purchased a dph-140s to test with AAH 1.5. I think this is a rebadged ACT P104SLD which others seem to have working with *. It seems to be configured and registered similarly to the softphones I've been using just fine, but it does not receive or send audio (it will send audio to vm), or perform the loopback test. It seems to signal and receive calls fine. The phone is on an internal lan with a local aah server. I'm merely guessing but it seems like the rtp side is not communicating properly. The codec selected is ulaw as in the config, I changed what they call media port from 41000 to 1, to no avail. Does anyone have a fully working dlink phone with aah, or particularly this model? It seems very close to working. The puzzing thing to me is you can never hear any audio (tones are ok), and the audio is heard in a vm message. The phone is pretty decent otherwise so any tips would be appreciated. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie install problem. And I alreadysearchedeverywhere!
Jeroen,
On 04-Sep-2005 19:34, Jeroen Baten wrote:
Op zondag 4 september 2005 11:10, schreef Stijn Jonker:
Hello Jeroen,
On 04-Sep-2005 9:22, Jeroen Baten wrote:
Op zondag 4 september 2005 09:07, schreef Jeroen Baten:
Op zondag 4 september 2005 00:21, schreef [EMAIL PROTECTED]:
Ok - what the heck is this?v
Dial(SIP/4001-2fea, Zap/2/mycellphonenumber})
I've never seen a } used in a dial statement...
sorry, one } to many. in extensions it reads:
exten = _97.,1,Dial(Zap/2/${EXTEN:2})
still no joy though. Even changed the timeout:
Executing Dial(SIP/4001-b455, Zap/2/456343|15) in new stack
-- Called 2/456343
-- Channel 0/2, span 1 got hangup
-- Hungup 'Zap/2-1'
== No one is available to answer at this time
What ISDN card do you have? It sounds like a quadbri, and that your in
the netherlands, so am I. That leaves one question is it directly
connected to a kpn line or via an other pbx/pabx?
I have a BRI 2-channel Eicon Diva on KPN and yes, look at the sig, I'm dutch.
Which part of your conf should I use and not use?
That depends on the driver, do you use chan_capi or junghanns bristuff
for the eicon?
P.S. That you where dutch I figured out, but looking at i2rs.nl you
deliver services so it could have been in germany or so... ;-) Never
expect the usual, always expect the unusual ;-)
Stijn
--
Met Vriendelijke groet/Yours Sincerely
Stijn Jonker [EMAIL PROTECTED]
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Re: [Asterisk-Users] newbie install problem. And I alreadysearchedeverywhere!
On Sun, 4 Sep 2005, Stijn Jonker wrote:
On 04-Sep-2005 19:34, Jeroen Baten wrote:
Op zondag 4 september 2005 11:10, schreef Stijn Jonker:
Hello Jeroen,
On 04-Sep-2005 9:22, Jeroen Baten wrote:
Op zondag 4 september 2005 09:07, schreef Jeroen Baten:
Op zondag 4 september 2005 00:21, schreef [EMAIL PROTECTED]:
Ok - what the heck is this?v
Dial(SIP/4001-2fea, Zap/2/mycellphonenumber})
I've never seen a } used in a dial statement...
sorry, one } to many. in extensions it reads:
exten = _97.,1,Dial(Zap/2/${EXTEN:2})
still no joy though. Even changed the timeout:
Executing Dial(SIP/4001-b455, Zap/2/456343|15) in new stack
-- Called 2/456343
-- Channel 0/2, span 1 got hangup
-- Hungup 'Zap/2-1'
== No one is available to answer at this time
What ISDN card do you have? It sounds like a quadbri, and that your in
the netherlands, so am I. That leaves one question is it directly
connected to a kpn line or via an other pbx/pabx?
I have a BRI 2-channel Eicon Diva on KPN and yes, look at the sig, I'm
dutch.
Which part of your conf should I use and not use?
That depends on the driver, do you use chan_capi or junghanns bristuff
for the eicon?
bristuff for Eicon Diva card? That's not possible.
Which card do you use exactly? Is it a DIVA PCI or DIVA Server card?
In case of DIVA PCI you can use mISDN/chan_misdn. For DIVA Server
chan_capi would be your choice.
Armin
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[Asterisk-Users] OT: Sipura SPA 200 Caller ID Problem
Sorry to bug all of you with this, but I have to assume there are a number of Sipura experts here... I have a Sipura SPA 2000 that I've been using for nearly 2 years now. It's flashed up to firmware 3.1.5. On line 1, I no longer get Caller ID (it used to work, and I can't remember when it stopped). On line 2, I always get Caller ID. To my old eyes, _every_ switch on both lines seems configured identically. I can see that calls to Line 1 have the correct Caller ID on the Asterisk CLI, and it displays correctly on my soft phone, so the problem is definitely at the Sipura level. As a hint, I think that I was playing with some attended transfer settings in Asterisk (not on the Sipura), and I may have typed one of the magical *XX codes the went to the Sipura instead of to Asterisk. Anyway, I've tried to type in each one that looked like a candidate for affecting this, and nothing seems to work. Any pointers would be greatly appreciated. Thanks in advance! P.S. I reset the Sipura to factory defaults, and rebuilt from there. It still works on line 2 and not on line 1 :-(. It's not the phone which is a Uniden Tru8866, since all handsets exhibit the same problem... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP, NAT and MySQL support (sipfriends)
Hi all, I am new to asterisk and I can not find any detailed info on using SIP MySQL support (sipfriends) with clients behind NAT. I've heard that I have to patch chan_sip.c and Makefile to get it working. I tried on voip-info.org but found no answer for my questions. I found some answer on Digium mail list archive: http://lists.digium.com/pipermail/asterisk-cvs/2004-January/000854.html but I do not know for what version of Asterisk it is aplicable. I hope you will help me a bit. -- Jarek ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sipura spc.exe ?
Can someone send me a copy of the sipura spc.exe used to compile centralized configs via private email? I've applied for their access but haven't been approved just yet. Need to do some testing with it over the next few days. Rich ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Option 1 in IVR menu
Hi all, I'm trying to setup a simple IVR menu in a context in extensions.conf. So far, I have: extension s for playing back the menu # to repeat it * for directory 0 for operator 1 which goes to another context: exten = 1,1,GoTo(option_1,s,1) Here is what I have in extensions.conf: [incoming] ; main greeintg exten = s, 1, Ringing exten = s, 2, Wait(10) exten = s, 3, NoOp() exten = s, 4, Answer exten = s, 5, Playback(silence/1) exten = s, 6, Background(a2) exten = s, 7, Background(b) exten = s, 8, WaitExten(20) exten = s, 9, Hangup ; repeats the message exten = #,1,Goto(s,6) ;Operator exten = 0,1,Macro(stdexten,302) ; Dial extension exten = 1,1,Goto(option_1,s,1) ; this one is delayed exten = 2,1,Goto(option_1,s,1) ; this one is executed immediately ;Directory exten = *,1,Directory(default) exten = t,1,Goto(s7) exten = i,1,Playback(invalid) The problem is that when user presses 1, there's a delay before the action Goto is executed (I'm assuming delay is caused by some timeout). All other options work fine, if I use 2 or any other number instead of 1 for that action, it also works fine - the Goto gets executed immediately. I do get the standard NOTICE[11563]: rtp.c:281 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. when user presses a key. So what is so special about sending DTMF key 1 and how can I get around it? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Messagenet.it
Hi to all, I need help to setting up messagenet.it account in Asterisk. My * is connected with static IP to the net. No other cards at the moment, just the network-card. I'm able to receive call on the geographical number, but I'm not able to setup the outgoing calls. All that it does is: I dial the number(not voip), * accepts my call, I can hear the ring tone, but the telephone called do not receive the call. I'm sure, because I called my number and I've picked up the phone while is was ringing, and the call was not there... I'm using port 5061 and rtp port 8000. Username and password are correct. I'm sure because I've setup a voip phone with those and it worked. Any ideas? Thanks a lot-- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Option 1 in IVR menu
On Sun, 2005-09-04 at 14:09 -0700, Adrian A wrote: Hi all, I'm trying to setup a simple IVR menu in a context in extensions.conf. So far, I have: extension s for playing back the menu # to repeat it * for directory 0 for operator 1 which goes to another context: exten = 1,1,GoTo(option_1,s,1) Here is what I have in extensions.conf: [incoming] ; main greeintg exten = s, 1, Ringing exten = s, 2, Wait(10) exten = s, 3, NoOp() exten = s, 4, Answer exten = s, 5, Playback(silence/1) exten = s, 6, Background(a2) exten = s, 7, Background(b) exten = s, 8, WaitExten(20) exten = s, 9, Hangup ; repeats the message exten = #,1,Goto(s,6) ;Operator exten = 0,1,Macro(stdexten,302) ; Dial extension exten = 1,1,Goto(option_1,s,1) ; this one is delayed exten = 2,1,Goto(option_1,s,1) ; this one is executed immediately ;Directory exten = *,1,Directory(default) exten = t,1,Goto(s7) exten = i,1,Playback(invalid) The problem is that when user presses 1, there's a delay before the action Goto is executed (I'm assuming delay is caused by some timeout). All other options work fine, if I use 2 or any other number You must have additional lines in that context that you haven't shown us, therefore we can't really help you. In any case, it is because some other extension *might* be matched, so asterisk is waiting for more digits, after the timeout, it decides to just process what it has, and follows your goto. ie, you have something like this: exten = 1,1,Goto(context,s,1) exten = 1123,1,Playback(test) Remember that even included contexts will be looked into... and also there are patterns that might match like: exten = _1X.,1,Playback(test) Regards, Adam ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
FW: [Asterisk-Users] OH323 with [EMAIL PROTECTED] - seems incomplete
Thank you (for spamming) - it was the clue I needed to push this through. Sorry it took me a while (and a google :-) ) to realize you'd addressed my initial query - basically, my loss. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Becker Sent: Wednesday, August 24, 2005 00:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OH323 with [EMAIL PROTECTED] - seems incomplete Abdel Rahman Tarzi wrote: I installed oh323 and everything seemed to go smoothly (compile everything upto calling through using oh323). I must admit, there is some behavior that's doesn't seem right but generally, I'm able to dial-out of any oh323 device whether to an extension or to a trunk. Audio is sometimes muted when dialing out until the extension or dialed number answers. Sound quality is good when it's there. Following the install, I do not seem to have the option to create an oh323 extension or trunk. Something that I need to do. I realize it's possible to edit the .conf but I needed to ask whether this was normal - doesn't seem like it is to me. Naturally, I'm apprehensive that editing the .conf files manually may be overwritten by AMP, but even if not, it would seem like it should've been the norm to have added an entry like oh323 to SIP, IAX2 etcetera types of trunks. Also, extension types (which work from a drop-down list) should have been modified. I would appreciate if someone familiar with the application could inform whether this is normal before I attempt to add extensions and trunks manually. I apologize for spamming the list... I failed to mention that Custom Trunks support exists in the current version of AMP (1.10.008). Here is the text from the Custom Dial String tooltip: Define the custom Dial String. Include the token $OUTNUM$ wherever the number to dial should go. examples: CAPI/:b$OUTNUM$,30,r H323/[EMAIL PROTECTED] OH323/[EMAIL PROTECTED]: vpb/1-1/$OUTNUM$ -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Eyebeam
Hello, What's the status on using eyebeam with Asterisk, does it still require a patch to Asterisk to support the video component? I'm intererested in starting to use Video and audio telephony but wary of anything that requires patches. cvs head works out of the box, just enable the h.323+ codec. cvs head is what will become asterisk 1.2, check for betas... Juraj. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Nokia 32 Terminal
I use a Nokia 32 as an extension and as a trunk (two sites).. Which are you doing ? If you wish to connect it to an FXS you will need a special cable which Nokia sells.. Connecting to an FXO (which expects a line) is the default. Check the normal stuff (like dialstring) before you suspect the device.. They're really maintenance-free !! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of andrutto Sent: Sunday, September 04, 2005 22:14 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Nokia 32 Terminal Hi, Does anyone have some experience with Nokia 32 Terminal (it is an analog GSM Gateway)? After a configuration I can make only incoming calls, I'm not able to do any outgoing. Nokia signalize an error (4 short tones), when I try to phone someone. I tried postpaid simcards as well as prepaid simcards with the same result. Does anyone try to connect this gateway to Asterisk PBX if so what were the results? All the best Andrutto -- Jedyny taki czat... http://link.interia.pl/f18b0 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: [EMAIL PROTECTED] - requesting help on oh323, ISDN BRI and iConnectHere DID
Thanks again. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Becker Sent: Sunday, September 04, 2005 18:20 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FW: [EMAIL PROTECTED] - requesting help on oh323,ISDN BRI and iConnectHere DID AbdelRahman Tarzi wrote: I know almost nothing linux, and don't really know that much about Asterisk (proper).. but I was 'pulled' by this subject and grabbed an [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] installation CD (version 1.3) and just went with it. Newbie doesn't quite describe it, I'm a banker.. this simply goes to show how easy Asterisk is becoming (I certainly hope this direction was meant to be inviting to people like me). I've been at this for a little over three weeks now. I've researched these topics but have not found satisfactory answers to the following: If there are places where an answer could be found, I'd appreciate some pointer(s). Please post to the [EMAIL PROTECTED] forum: http://sourceforge.net/forum/forum.php?forum_id=420324 and/or amportal mailing list: http://lists.sourceforge.net/lists/listinfo/amportal-users or Help forum: http://sourceforge.net/forum/forum.php?forum_id=414452 oh323 Once it (oh323) was installed, I was able to dialout from an h323 device with little problem. I have not, however, been able to setup the h323 device as an extension (or use it to communicate with other h323 devices, or through it to the pstn.. I was confused because in creating an extension (or a trunk) in [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] there was nothing to indicate that h323 was available. The dropdown in extension creation does not include oh323 (or some such) item. I need a clue as to how to setup an extension and how to setup a trunk (using an h323 device that is). The current version of AMP (1.10.008) supports the creation of Custom Trunks, which in your case could be an oh323 trunk. The next version of AMP (1.10.009, currently in second beta) will support Custom extensions, which in your case would mean that you could setup your oh323 endpoint and be able to leverage AMP's dialplan. ISDN Could someone please tell me whether the AVM Fritz card is a low pain solution to connecting to a BRI ? I've tried connecting an ASUS card (doesn't work with the available drivers/application) so I'm really asking in order to buy. I only have one BRI - this is at home) so would appreciate any help. Can't comment on the pain aspect but documentation on this configuration exists: http://www.voip-info.org/tiki-index.php?page=Asterisk+AVM+Fritz+CAPI+Driver+ Install (Note: this would be configured in AMP as a Custom Trunk as well) iConnectHere From day one, I was able to create the trunk to dialout of iConnectHere, but despite finding several claimed correct settings for receiving its DID, I've not been able to. When I connect a Grandstream 101 to the line and power it up, it has absolutely no problem receiving calls from that DID but no matter what I try, I'm unable to receive into Asterisk (@home) .. Just so we're clear, I am receiving calls from fwd and through fwd from an ipkall number.. and I'm also recieving calls from the FXO on a sipura spa3k.. - iConnectHere don't help more than point to the wiki .. (and what I find there doesn't work). No comment. Never heard of this VoIP Service Provider. Just a note to describe my handicap with linux: I'm unable to capture a log (or perhaps it's captured and I'm just not aware where).. Asterisk: /var/log/asterisk/full (see /etc/asterisk/logger.conf) Linux: /var/log/messages [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] seems to dialout and receive calls using Macros.. I suspect it's a clever way of managing the setup, but I'm not sure where the various portions of SIP.conf, extensions.conf, extensions_additional.conf, extensions_custom.conf or indeed oh323.conf. - are relevant. Please search the [EMAIL PROTECTED] forum and/or amportal list forum for more info on these topics. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Updated Chan Unistim?
Hi, Does anybody have an updated Chan Unistim that compiles on Asterisk 1.2beta? Below is the output when compiling on Red Hat 9.0 Thanks, [EMAIL PROTECTED] chan_unistim-0.9.2]# make gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_unistim.o chan_unistim.c chan_unistim.c:17:34: asterisk/channel_pvt.h: No such file or directory chan_unistim.c: In function `SendClient': chan_unistim.c:613: warning: implicit declaration of function `sendto' chan_unistim.c: In function `unistim_ss': chan_unistim.c:1332: dereferencing pointer to incomplete type chan_unistim.c:1344: structure has no member named `callerid' chan_unistim.c:1345: structure has no member named `ani' chan_unistim.c: In function `KeyCall': chan_unistim.c:1633: structure has no member named `bridge' chan_unistim.c:1637: structure has no member named `bridge' chan_unistim.c:1643: structure has no member named `bridge' chan_unistim.c: In function `unistim_call': chan_unistim.c:2358: dereferencing pointer to incomplete type chan_unistim.c:2386: structure has no member named `callerid' chan_unistim.c:2388: structure has no member named `callerid' chan_unistim.c:2388: structure has no member named `callerid' chan_unistim.c:2388: structure has no member named `callerid' chan_unistim.c:2388: structure has no member named `callerid' chan_unistim.c:2388: structure has no member named `callerid' chan_unistim.c: In function `unistim_hangup': chan_unistim.c:2413: dereferencing pointer to incomplete type chan_unistim.c:2419: dereferencing pointer to incomplete type chan_unistim.c:2426: dereferencing pointer to incomplete type chan_unistim.c: In function `unistim_answer': chan_unistim.c:2463: dereferencing pointer to incomplete type chan_unistim.c: In function `unistimsock_read': chan_unistim.c:2497: warning: implicit declaration of function `recvfrom' chan_unistim.c: In function `unistim_read': chan_unistim.c:2588: dereferencing pointer to incomplete type chan_unistim.c: In function `unistim_write': chan_unistim.c:2598: dereferencing pointer to incomplete type chan_unistim.c: In function `unistim_fixup': chan_unistim.c:2636: dereferencing pointer to incomplete type chan_unistim.c: In function `unistim_indicate': chan_unistim.c:2687: dereferencing pointer to incomplete type chan_unistim.c: In function `unistim_sendtext': chan_unistim.c:2807: dereferencing pointer to incomplete type chan_unistim.c: In function `unistim_new': chan_unistim.c:2889: dereferencing pointer to incomplete type chan_unistim.c:2891: dereferencing pointer to incomplete type chan_unistim.c:2892: dereferencing pointer to incomplete type chan_unistim.c:2893: dereferencing pointer to incomplete type chan_unistim.c:2894: dereferencing pointer to incomplete type chan_unistim.c:2895: dereferencing pointer to incomplete type chan_unistim.c:2896: dereferencing pointer to incomplete type chan_unistim.c:2897: dereferencing pointer to incomplete type chan_unistim.c:2898: dereferencing pointer to incomplete type chan_unistim.c:2899: dereferencing pointer to incomplete type chan_unistim.c:2900: dereferencing pointer to incomplete type chan_unistim.c:2901: dereferencing pointer to incomplete type chan_unistim.c:2921: structure has no member named `callerid' chan_unistim.c: In function `reload_config': chan_unistim.c:3763: warning: implicit declaration of function `socket' chan_unistim.c:3771: warning: implicit declaration of function `setsockopt' chan_unistim.c:3772: warning: implicit declaration of function `bind' chan_unistim.c: In function `unistim_get_rtp_peer': chan_unistim.c:3806: dereferencing pointer to incomplete type chan_unistim.c: In function `unistim_set_rtp_peer': chan_unistim.c:3818: dereferencing pointer to incomplete type chan_unistim.c: In function `load_module': chan_unistim.c:3892: warning: passing arg 1 of `ast_channel_register' from incompatible pointer type chan_unistim.c:3892: too many arguments to function `ast_channel_register' chan_unistim.c:3900: warning: assignment of read-only member `type' chan_unistim.c: In function `__unload_module': chan_unistim.c:3923: warning: passing arg 1 of `ast_channel_unregister' from incompatible pointer type make: *** [chan_unistim.o] Error 1 -- Andres ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Real-Time Voicemail Configuration
Hi, When using asterisk real-time with mysql voicemail integration... where exactly do I put the options like the [PBX] tag, and how long silence can be, etc? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: ztdummy and Linux 2.6.13-rc7
Tony Mountifield wrote: In article [EMAIL PROTECTED], Doug Lytle [EMAIL PROTECTED] wrote: Anybody having issues with ztdummy under the current 2.6 RC7? I get the following errors when trying to modprobe ztdummy: Unable to register zaptel rtc driver Doing a Google on the error shows reference to a message from 2004 that said you might not have RTC compiled into the kernel. Checking via: cd /usr/src/linux-2.6.13-rc7 grep -i rtc .config shows: CONFIG_APM_RTC_IS_GMT=y CONFIG_RTC=m CONFIG_GEN_RTC=m CONFIG_GEN_RTC_X=y CONFIG_HPET_RTC_IRQ=y CONFIG_SENSORS_RTC8564=m CONFIG_SND_RTCTIMER=m Any suggestions? rtc and genrtc are alternatives to each other. Make sure that the rtc module is loaded, and *not* genrtc. ztdummy is not compatible with genrtc, only with rtc. I had time tonight to try this. Under Linux 2.6.13 final. Looking at make menuconfig shows that both Generic /dev/rtc emulation and Enhanced Real Time Clock support Removing one and enabling the other, compiling and recompiling zaptel: make clean;make linux26 make install (udev rules in place) I am unable to do a modprobe ztdummy without the above error. Any others running Linux 2.6.13 and successfully using ztdummy for timing? Doug ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk Real-Time Voicemail Configuration
I should add to this... I understand to make the table.. but when I make it.. asterisk selects it but seems to ignore things. No where have I found documented what the var_category and such are... what numbers do I put in there?!?! On 9/4/05, Matt [EMAIL PROTECTED] wrote: Hi, When using asterisk real-time with mysql voicemail integration... where exactly do I put the options like the [PBX] tag, and how long silence can be, etc? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Rich Adamson wrote: Sorry to write so many consecutive messages in such a short period of time, but this problem is really bugging me as it has been going on for days. When I look in Ethereal, there are actually two calls going on -- in this particular call, Source call #4 and Source call #10318, #4 coming from the asterisk server and the other one coming from my computer to the Asterisk server. I don't know why there are two separate calls, but perhaps one of you do. Its not really two separate calls; its the transmit leg and the receive leg of the same call. Oh, ok, that makes sense. Anyhow, source call #10318 seems fine, sending a new packet every 20 ms pretty much perfectly and all (although I do see now that one packet has a timestamp of 33080 and the next has one of 35060 -- is this something to be concerned about? The diff is 1980, which essentially suggests there were 99 iax packets missing between those two timestamps. (Should note that timestamps miscalculations have been an issue in the iax2 source code, but I don't remember if some of the fixes were before or after the version of code your running. That really was the basis for suggesting a code upgrade.) If 99 iax2 packets are actually missing, you _would_ have a problem with the audio quality. Perhaps I read that wrong, as the different filters don't seem to show those packets as lost or jittered. Plus, this crackle happens fairly often, so I don't know if it's any indication of dropped packets (or the root of the problem, at least), but then again, I'm not sure. I will try taking another sample. it doesn't seem widespread). However, call #4 seems to send every 20 ms, but then there will be a pause or something in sending, in between which there will be more packets from source call #10318 which are sent pretty much OK. Keep in mind that you're looking at a full duplex flow of packets in and out. The fact that packets are not exactly one transmit for one every one received is not as important as identifying missing packets in the form of large jumps in timestamp values. Yeah, I see what you're saying. Then, the next packet for source call #4 will have a timestamp of something like 33540, exactly 200 ms after the previous packet from source call #10318. However, the next packet for SC (source call) #10318 increments 20 ms like it should. Every single packet then on (in this capture, I recorded about 1500 packets) sends perfectly. iax2.rrdropped, iax2.rrjitter, and iax2.iax.rrloss returned only 2 packets--the same two, in the middle of the 1500 packets. So, out of 1500, these are the only two that seem to have problems. Asterisk can generally handle a couple of missing 20 millisecond packets, but it can't compensate (or cover up) hugh timestamp jumps. Might consider doing another ethereal run or two and see if you can reach a conclusion that the timestamp jumps are in fact associated with the choppy audio. If you can reach that conclusion, the next step is to isolate why; is it bad code or the network that's causing the issue? I don't recall from your previous postings, but could you repeat what exact code versions are running on my computer and the remote asterisk server? By code versions, do you mean what OS I'm running on my computer? I'm running Debian etch (testing) and ethereal 0.10.12. On my Asterisk system, I'm also running Debian etch, with Asterisk 1.0.7 (debian's testing version of asterisk). Would you then suggest that I should upgrade to a later version? Which one? 1.0.9? Or are even the BETAs (1.2.x) usable? I'm not sure if it's just psychological at this point, but the crackle seems to have faded a bit; it doesn't sound as harsh or prevalent as it did before. I moved my power strip further away from my computer, and I'm not sure if this made a difference or not, but it doesn't sound *as bad*. Note that it's still bad enough that I would take a land line over it any day, and it's certainly and by all means abnormal (thus, absolutely worth resolving). I did listen to some sample sounds on Cisco's website -- http://www.cisco.com/en/US/tech/tk652/tk698/technologies_white_paper09186a00801545e4.shtml#crackle -- and determined that my symptom is most like their posted crackling audio quality symptom, if this helps any more. However, /their/ crackling example is much harsher, louder, and more annoying than mine. Nonetheless, I believe mine more closely resembles that sample than any of the others -- it's just less severe. It could be a big transformer that's near the Ethernet card, but I believe my strip is now a pretty standard distance away from my Ethernet card; I measured it, and it's roughly 16 inches from the Ethernet card itself. It doesn't seem like it will that much farther, but is that too close? Plus, if it is electrical interference, ethernet being digital and all, wouldn't the potential interference render
RE: [Asterisk-Users] Re: ztdummy and Linux 2.6.13-rc7
You need to care about the _actual_ error, not the report there is an
error. The error is (usually) reported to the console. Reboot the
computer and type this:
dmesg -c /dev/null
modprobe ztdummy
dmesg
The output of the second dmesg will show you exactly what the error
message is.
Being that you have 'hpet' enabled, it's going to be 'Input/Output
error'. There's code in the kernel rtc driver that doesn't let you use
it if hpet is on:
--snippy--
int rtc_register(rtc_task_t *task)
{
#ifndef RTC_IRQ
return -EIO;
#else
if (task == NULL || task-func == NULL)
return -EINVAL;
--snippy--
[From your kernel config]
CONFIG_HPET_RTC_IRQ=y
So. Turn that off, and recompile the rtc module and it'll start working
--Rob
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[Asterisk-Users] Snom hints/lights
I updated our system here to the latest CVS (we were previously running 25/08/05) and the lights work perfectly now. PaulH CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] kernel panic
I've just loaded zaptel 1.0.9 on a new 2.6.12 system (FC4 with updates). The system has a TE110P card, and zaptel.conf is configured for an E1. When I do a 'zaptel stop' I get a kernel panic. Has anyone else seen this? Thanks, ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Perhaps I read that wrong, as the different filters don't seem to show those packets as lost or jittered. Plus, this crackle happens fairly often, so I don't know if it's any indication of dropped packets (or the root of the problem, at least), but then again, I'm not sure. I will try taking another sample. Send me a copy of an ethereal trace off list and I'll take a look at it. When you do that, let me know what IP address is what in that trace so I have some clue what I'm looking at. Would you then suggest that I should upgrade to a later version? Which one? 1.0.9? Or are even the BETAs (1.2.x) usable? I'm not sure what the latest stable version is since I don't pay much attention to it. I stay with the cvs head keeping a backup copy of the previous working code on my system in case I have issues with whatever I check out. If v1.0.9 is the latest stable, then use it. You have kind of skipped over exactly what my computer happens to be. Specifically, tell us what O/S, what software are you running on that system that is communicating with iax2, etc. Also, when you refer to your Asterisk system, is that system on your local network or located somewhere else? It could be a big transformer that's near the Ethernet card, but I believe my strip is now a pretty standard distance away from my Ethernet card; I measured it, and it's roughly 16 inches from the Ethernet card itself. It doesn't seem like it will that much farther, but is that too close? If external electrical noise from a transformer is impacting your ethernet cable, it would impact your music and other things as much as it would iax2. So that's probably not an issue. Power strips by themselves do not generate electrical noise, so that's a non-issue anyway. BTW, just to reassure you all that my ethernet is fine, here are the results of an extremely fast (ping -i 0.0005) ping to the asterisk server: --- 192.168.2.7 ping statistics --- 10758 packets transmitted, 10758 received, 0% packet loss, time 28976ms rtt min/avg/max/mdev = 0.106/0.118/10.779/0.105 ms, pipe 2, ipg/ewma 2.693/0.118 ms i don't think my ethernet is flawed at all. i did several of those tests, by the way. Okay, send me a reasonable ethereal trace and I'll take a look at it. Rich ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: ztdummy and Linux 2.6.13-rc7
Rob Thomas wrote: So. Turn that off, and recompile the rtc module and it'll start working Thank you very much! Doug ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to hear.
Hi all, Has anyone had problems with not being able to hear callers and them not being able to hear you? And had any success on how to fix it? Our call centre staff are complaining that this is a continual problem. Appreciate any thoughts on this. Regards Jennifer Hales ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Real-Time Voicemail Configuration
You must store voicemail.conf using RealTime Static in order to use the options you have mentioned from database. -Matthew From: Matt [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sun, 4 Sep 2005 19:51:36 -0400 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Real-Time Voicemail Configuration Hi, When using asterisk real-time with mysql voicemail integration... where exactly do I put the options like the [PBX] tag, and how long silence can be, etc? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Real-Time Voicemail Configuration
How did you convert your voicemail.conf file into RT Static? Did you use the perl script? -Matthew From: Matt [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sun, 4 Sep 2005 20:37:34 -0400 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Asterisk Real-Time Voicemail Configuration I should add to this... I understand to make the table.. but when I make it.. asterisk selects it but seems to ignore things. No where have I found documented what the var_category and such are... what numbers do I put in there?!?! On 9/4/05, Matt [EMAIL PROTECTED] wrote: Hi, When using asterisk real-time with mysql voicemail integration... where exactly do I put the options like the [PBX] tag, and how long silence can be, etc? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Argentina - zapata.conf switchtype for Argentina
Guillermo, Switchtype depends on to which kind of PSTN are you connected to. Are you connected to Telecom or Telefonica?, using PRI or FXO/FXS lines? Normally both follows European Standards for Telephony (CCITT), not Bell standars. And in the case of Telecom they have a lot of Telettra equipment installed. I hope this can help you. Carlos Alperin [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guillermo Freige Sent: Sunday, September 04, 2005 12:03 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Argentina - zapata.conf switchtype for Argentina Probably you need to use unicall+mfcr2 support instead of zapata, as Argentina uses R2. Guillermo From: Leandro Rzezak [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Argentina - zapata.conf switchtype for Argentina Date: Sat, 3 Sep 2005 18:54:59 -0300 Just to receive a recommendation on switchtype for Argentina, Buenos Aires, 114816. Thanks a lot -- Leandro Rzezak [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Automon filenames
Guys. How are filenames determined for automon and queue recordings enabled on queues.conf? I see the names have some tomestamps or something but is there a way to predefine the filenames to use? Thx! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A few questions before final proposal...
I am attempting to assemble a proposal for a client of mine that is looking to replace their phone system. I think it's a good first installation with 4 POTS incoming and 15 extensions, with an overhead paging system. I also think that it would make a good case for OSS applications in general. Being new to * I have a few more questions, but won’t flood the list with them. I’m not new to Linux, but new to *. I am a bit hesitant to push * as a solution for this client, mostly due to my limited knowledge of * and its features, I am concerned that 1.) The phones I chose will not fit their needs, and 2.) * will not provide the features that they need. Things like shared line appearances. 3.) Phone Compatibility; The Bugetone 2000 has been highly recommended, but I'm still looking at Polycom (501) and Aastra phones (9133i). How are your experiences with these phones? Is there a better model for supporting advanced features? Features like FollowMe, Share Line Appearances, etc… Being my first install, should there be anything that I should keep in mind? I have read voip-info sections regarding rollouts, but as always, help from the community is greatly welcomed. What are the general thoughts on the O’Reilly book? ~kurth Kurth Bemis AIM: Evil Santabot ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sending fax
I've read alot on the wiki about sending and receiving faxes thru asterisk. I've gotten the receive to work great.My question is how does one send a fax? I see lots of instructions about how to send the image to asterisk by email, etc. The problem is how does one make the image of the fax to begin with? Has anyone come up with a good solution for this? Regards, Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A few questions before final proposal...
On Mon, 2005-09-05 at 01:31 -0400, Kurth Bemis wrote: I am attempting to assemble a proposal for a client of mine that is looking to replace their phone system. I think it's a good first installation with 4 POTS incoming and 15 extensions, with an overhead paging system. I also think that it would make a good case for OSS applications in general. Being new to * I have a few more questions, but won’t flood the list with them. I’m not new to Linux, but new to *. I am a bit hesitant to I would suggest that you find a local asterisk 'consultant' who is willing to help you get your feet wet. Maybe you can budget an extra couple of hundred dollars, but it will be worth it having someone to turn to for assistance. As far as handsets, I'd suggest the Polycom IP600 (since it has more line appearances that the IP500, and it just looks a little nicer). Otherwise, I've never used them, and I really like the polycom's, but I keep hearing that the snom phones handle shared call appearances really well, and they have more of them etc The only other issue I'd be wary of is all the echo issues associated with analog lines. Oh, and the overhead paging could get a bit tricky depending on the specific requirements (I've never done any paging at all, so I'm not so sure about that). So, from a feature point of view, asterisk can do most anything demanded of it, and usually so much more, but things like shared line appearances are easier using FOP than a led on a phone Regards, Adam ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
