Re: [asterisk-users] Queue autopause
On Thu, Jul 9, 2009 at 4:41 PM, Miguel Molinammol...@millenium.com.co wrote:
Christian Gansberger escribió:
On Thu, Jul 9, 2009 at 12:21 AM, Miguel Molinammol...@millenium.com.co
wrote:
Christian Gansberger escribió:
Hi all!
I want to autopause my queue member when they are not answering within
20 seconds, and the autopause
should affect all queues they are member of, not only the queue where
the call was not answered.
Is there a way to do that?
The members gets dynamically added. I'm using asterisk 1.4.21.2.
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Why would you want to do that? The purpose of the autopause is to
discard the absent agent that is not responding to the calls to not
try it anymore until it gets unpaused by a supervisor or someone else,
and therefore the pause is made to all queues the agent is member of.
Why pause it on only one queue, letting it ring on other queues?
Aside from the purpose you have on this, I think you would need to
modify the app_queue.c code to make the parameter configurable inside
each queue definition and not on the general section of queues.conf.
Then you would need to modify the logic to handle the autopause
configured for each queue. This is a general idea as I didn't take a
deep look of app_queue.c to see how it works exactly.
Any other solution without changing asterisk code would imply a external
application that monitors the queues and makes the custom autopause you
need.
Just my two cents...
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
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To make things clearer:
I want the queue member is autopaused on all queues. As a matter of
fact in asterisk (vers. 1.4.24.1)
the queue member is only paused on one Queue.
I tried setting autopause=yes in general context, which doesn't do anything.
So i set autopause=yes in every Queue definition, which is working,
but only on that queue.
I don't use the agents channel (well i tried, with ending up in lots
of trouble), because its
depreciated in asterisk 1.4 and gone in 1.6. so i decided to
do as proposed in UPGRADE.txt and
asterisk-src/doc/queues-with-callback-members.txt,
with one change, i'm not using the Local channel, because it is not
showing the right status
of the devices in the queue. (I wonder how the callcenter at digiums
ist working with that).
maybe anyone else having problems with queues in asterisk 1.4?
yours
christian gansberger
You're right, the autopause on its standard behavior pauses only the member
of the queue where it belongs. Taking a little look at app_queue.c
(http://www.asterisk.org/doxygen/1.4/app__queue_8c.html) you can very easily
patch the source code to achieve the functionality you want. The key
functions are:
static int set_member_paused - Traverses the queues doing all the things
necessary on all different scenarios (realtime, etc) to pause the member you
give to it. If there's no queue name given, it with pause the member on all
queues (the PAUSEALL event).
static void rna - (as the doxygen doc says) RNA == Ring No Answer. Common
code that is executed when we try a queue member and they don't answer.
If you take a look to the rna function, with autopaused enabled it will
pause the member if it doesn't answer the queue call after the timeout time.
You can make it pause all members just by changing this one line:
02164 if (!set_member_paused(qe-parent-name, interface, 1)) {
to
02164 if (!set_member_paused(, interface, 1)) {
That way we don't send the queue name, pausing it in all the queues it is
member of.
Although it's not tested, it might work for you. That's the beauty of
Asterisk and well documented Open Source projects, you can get to the code
as deep as you want, learn from it how it works, and change it/improve it
according to your needs. Good contributions make it to the official code as
well.
Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
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Thanks!
I will try changing the app_queue, on my testsystem in the next days.
For now I made some dummy-queues (timeout=1), and one queue where all
the calls are taking place
and the members are logged on.
Cheers
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Re: [asterisk-users] PRI failover to SIP trunk
On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote:
You have a small typo:
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten = _.,2,Dial(SIP,Provider,${EXTEN})
exten = _.,1,Dial(Zap/g1/${EXTEN})
exten = _.,2,Dial(SIP/Provider/${EXTEN})
('/' instead of ',')
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Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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Re: [asterisk-users] [Asterisk-users] SendFAX/T.38 question
How about:
Exten = _X.,3,Dial(SIP/${ext...@carrier,60,M(fax-out))
[macro-fax-out]
exten = s,1,Set(FAXFILE=/root/test.tif)
exten = s,2,Set(LOCALHEADERINFO=WHO CARES WHO I AM ?)
exten = s,3,Set(LOCALSTATIONID=1-800-Who-CARES)
exten = s,4,SendFax(${FAXFILE})
- Original Message -
From: jonathan augenstine
To: asterisk-users@lists.digium.com
Sent: Saturday, March 14, 2009 09:12
Subject: [asterisk-users] [Asterisk-users] SendFAX/T.38 question
I have some questions about the T.38 faxing capability. I have been able to
successfully setup the inbound receive fax. However, I am having problems
tracking down the format of the outbound extensions.conf SendFAX command. I
have looked at the code and it looks like it only takes a single parameter, a
file name. But the attempts I have tried seem unsucessful. I have tried
dialing out and then calling SendFAX and calling SendFAX before the dial. No
success.
Can someone please provide me with an extensions.conf example of how to use
SendFAX?
Thank you.
Jonathan Augenstine
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[asterisk-users] Friday July 10th: Gigaset DECT/SIP phones have come to the USA
Hi, This week Tony Stankus, North American product manager of the Gigaset line is our guest on VoIP Users Conference. I have had a two handset S675IP in our small business for about a year now and my wife and I both like the phone. But as a geek, I like it a lot more than she does :) 6 SIP lines avails for all those ITSP accounts I have, plus a regular PSTN. Vmail works on both, so if you have DID without voice mail service, your local Gigaset will handle the SIP channel as if it were a ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Friday July 10th: Gigaset DECT/SIP phones have come to the USA
(thank you gmail) so if you have DID without voice mail service, your local Gigaset will handle the SIP channel as if it were a PSTN line. This feature is selectable on a per account basis. The phones also do g722 so they work with our ZipDX wideband bridge. If you are considering new DECT phones for use with SIP, the line is something you should look at. Come and ask questions Friday at 12 Noon EDT on the VUC: IRC: #voip-users-conference SIP 7463#2262...@proxy.ideasip.com g722 SIP: 200...@login.zipdx.com or see http://VUC.me for more info See you at Astricon 2009 in October ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Source for OpenVox cards?
Hi I am in Canada and I finally went a head with http://www.cigear.com. It was quick painless and I did not have to deal with the customs and duties... Tim On Tue, Jul 7, 2009 at 5:54 AM, Tom O'Connor t...@twinhelix.org wrote: On Mon, Jul 6, 2009 at 5:03 PM, Tony Mountifield t...@softins.clara.co.uk wrote: In article be95b6c50907050425h777a9eha25924d88b5ba...@mail.gmail.com, Timothy Legge timle...@gmail.com wrote: I am looking for a source for an OpenVox card. Has anyone purchased through http://www.voiplink.com or do you normally use another vendor or OpenVox.cn directly? thanks Tim I have used voipon.co.uk, but I don't know whether that's useful to you, as you didn't say which country you are in. Cheers Tony Yes, I've used voipon.co.uk also, their customer service is very efficient. They'd dispatched the cards within 2 hours of me ordering them. -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using a mobile phone via USB as an extension
2009/7/9 Nick Hill t...@nickhill.co.uk Hello Sasa Carlos indicates that USB support may be available in chan_mobile but I can't find any references to it, and I think Oliver is looking for more info as well. That's perfectly true : I'm looking for more info for USB connectivity as chan_mobile seems silent about that (just as Google). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Kate AEL syntax ?
Hi, Is there something available to add AEL2 syntax highlighting support to Kate ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connecting two Asterisk together via SIP + DISA
Hi all, I need to test the following scenario: +---+ +---+ | asterisk 1| | asterisk 2| +---+ +---+ | | | | ___|__|___ | | | | | | +---+ +---+ | ATA 1 | | ATA 2 | +---+ +---+ / \ / \ /\ /\ 21 22 1011 That is, I have 2 asterisks connected via SIP, two ATAs with two lines, and the ATA1 is registered with asterisk1 and ATA2 is registered with asterisk2, and all incoming calls in asterisk2 from the asterisk1 (via SIP), are answered by a DISA. I can make calls between ATA1 and ATA2 without problems (the call will be routed to the asterisk1 to asterisk2, falls in DISA and I call one of the phones ATA2). I am now trying to make the call coming from,eg, extension 21, go to the asterisk1 - asterisk2, answered by the DISA and go back asterisk1, ringing the branch 22. Since I am newbie in this matter, I wonder with friends from the list if this is possible ... Or is there another way to do this Below is my conf files. Rgs Cesar === asterisk 1 ** sip.conf [21] type=friend context=phones ; Where to start in the dialplan when this phone calls secret=21 ;callerid=John Doe 1234 ; Full caller ID, to override the phones config ; on incoming calls to Asterisk host=dynamic; we have a static but private IP address ; No registration allowed ;nat=no ; there is not NAT between phone and Asterisk ;canreinvite=yes; allow RTP voice traffic to bypass Asterisk ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time ; from the phone to asterisk ; 1 for the explicit peer, 1 for the explicit user, ; remember that a friend equals 1 peer and 1 user in ; memory ; This will affect your subscriptions as well. ; There is no combined call counter for a friend ; so there's currently no way in sip.conf to limit ; to one inbound or outbound call per phone. Use ; the group counters in the dial plan for that. ; ;mailbox=1...@default ; mailbox 1234 in voicemail context default disallow=all ; need to disallow=all before we can use allow= allow=ulaw ; Note: In user sections the order of codecs ; listed with allow= does NOT matter! allow=alaw allow=g723.1 ; Asterisk only supports g723.1 pass-thru! allow=g729 ; Pass-thru only unless g729 license obtained ;callingpres=allowed_passed_screen; Set caller ID presentation ; See doc/callingpres.txt for more information [22] type=friend context=phones ; Where to start in the dialplan when this phone calls secret=22 ;callerid=John Doe 1234 ; Full caller ID, to override the phones config ; on incoming calls to Asterisk host=dynamic; we have a static but private IP address ; No registration allowed ;nat=no ; there is not NAT between phone and Asterisk ;canreinvite=yes; allow RTP voice traffic to bypass Asterisk ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time ; from the phone to asterisk ; 1 for the explicit peer, 1 for the explicit user, ; remember that a friend equals 1 peer and 1 user in ; memory ; This will affect your subscriptions as well. ; There is no combined call counter for a friend ; so there's currently no way in sip.conf to limit ; to one inbound or outbound call per phone. Use ; the group counters in the dial plan for that. ; ;mailbox=1...@default
Re: [asterisk-users] Kate AEL syntax ?
I have a basic config for AEL syntax highlighting for Kate if you would like it. - Brad From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Friday, July 10, 2009 8:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Kate AEL syntax ? Hi, Is there something available to add AEL2 syntax highlighting support to Kate ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
the “sip show peers command returns Name/username HostDyn Nat ACL Port Status 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline] I ran grep on the sip.conf and it didn't find any IPs. Where would I add my IP? I am guessing that the phones will not work for eternal calling if my SIP trunk is not configured correctly. I have had trouble finding the correct settings for the SIP trunk. I am still learning the termanology which has been part of my problem. From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Thu, 9 Jul 2009 16:17:07 -0500 Subject: Re: [asterisk-users] setting up phones What do you get from “sip show peers” in CLI? Do you have your ip address in sip.conf? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Thursday, July 09, 2009 4:12 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf [501] username = 501 transfer = yes mailbox = 501 call-limit = 100 type = peer fullname = 501 registersip = no host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 501 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = 501 nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 disallow = all allow = ulaw,gsm macaddress = 00085d10927f autoprov = yes label = 501 linenumber = 1 LINEKEYS = 1 [500] username = 500 transfer = yes mailbox = 500 call-limit = 100 type = peer fullname = 500 registersip = no host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 500 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = 500 nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 macaddress = 00085d1095aa autoprov = yes label = 500 linenumber = 1 LINEKEYS = 1 From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Thu, 9 Jul 2009 14:03:50 -0500 Subject: Re: [asterisk-users] setting up phones It should be pretty simple. Follow the instructions on this page http://www.voiptalk.org/products/aastra-setup.html put the username from sip.conf into the first 4 fields, the secret into the password field and your asterisk ip into the fields that say voiptalk.org users.conf [207] username=207 transfer=yes mailbox=207 call-limit=2 fullname=mickey mouse registersip=no host=dynamic callgroup=1 context=default cid_number=207 hasvoicemail=yes vmsecret=1234 email=u...@yourpbx.com threewaycalling=yes hasdirectory=yes callwaiting=yes hasmanager=yes managerread=system,call,log,verbose,command,agent,user,config managerwrite=system,call,log,verbose,command,agent,user,config hasagent=yes hassip=yes hasiax=no secret=x nat=yes canreinvite=no dtmfmode=rfc2833 insecure=no pickupgroup=1 macaddress=001170 autoprov=yes label=207 linenumber=1 disallow=all allow=ulaw,gsm sip.conf [207] type=peer context=phones host=dynamic fromuser=207 call-limit=3 secret=x canreinvite=yes directrtpsetup=no nat=yes qualify=yes register = 207:xx...@yourpbx.com/207 defaultip=1.2.3.4 mailbox=207 disallow=all allow=alaw From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Thursday, July 09, 2009 1:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] setting up phones Can someone tell me how to setup a Aastra 75i phone? I have been trying to set it up and have pointed it to our asterisk server and selected http for download. What is the path? I have created two extension in asterisk for testing. I can't even get the phones to call each other. Lauren found her dream laptop. Find the PC that’s right for you. Windows Live™: Keep your life in sync. Check it out. _ Lauren found her dream laptop. Find the PC that’s right for you. http://www.microsoft.com/windows/choosepc/?ocid=ftp_val_wl_290___ -- Bandwidth and Colocation Provided
Re: [asterisk-users] setting up phones
Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) _ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Video Call
Hi, I have 2 asterisk servers link via IAX. and i'm trying to do a video call. if 2 sip users are registered on the same server, the video works fine. but if 1 sip user is on server 1 and sip user 2 on sip server 2. there's no video at all. is it because call from sip server 1 goes to sip server 2 via IAX? hope my question is clear enough, thanks in advanced Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
Let's draw this out and let you fill in the blanks. Your asterisk server has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. Sip.conf should look this [phone1] type=peer context=phones host=dynamic fromuser=phone1 secret=secret1 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone1:secr...@foobar.com/phone1 defaultip=192.168.23.2 mailbox=1001 disallow=all allow=alaw [phone2] type=peer context=phones host=dynamic fromuser=phone2 secret=secret2 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone2:secr...@foobar.com/phone2 defaultip=192.168.23.3 mailbox=1002 disallow=all allow=alaw assuming your phones are set up to contact 192.168.23.1 with username phone1/phone2 and proper secret, all should register and you should be good to go. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 8:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. _ Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) _ Windows LiveT: Keep your life in sync. Check it out. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
Who is the carrier? What flavor of Asterisk are you using? Regardless, the phones should register and be able to call each other and other Asterisk apps if you have them in the dialplan. If you go to the Asterisk CLI and turn on SIP debugging, do you get anything at all? also, change registersip to yes. Thanks, Steve On Fri, Jul 10, 2009 at 9:32 AM, Ott Rose sixfourimp...@hotmail.com wrote: Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. -- Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.comwrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -- Windows Live™: Keep your life in sync. Check it out.http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
Asterisk registers with the phones? Obviously I have zero experience with these sets, but that is a new one. Thanks, Steve On Fri, Jul 10, 2009 at 9:40 AM, Danny Nicholas da...@debsinc.com wrote: Let’s draw this out and let you fill in the blanks. Your asterisk server has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. Sip.conf should look this [phone1] type=peer context=phones host=dynamic fromuser=phone1 secret=secret1 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone1:secr...@foobar.com/phone1 defaultip=192.168.23.2 mailbox=1001 disallow=all allow=alaw [phone2] type=peer context=phones host=dynamic fromuser=phone2 secret=secret2 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone2:secr...@foobar.com/phone2 defaultip=192.168.23.3 mailbox=1002 disallow=all allow=alaw assuming your phones are set up to contact 192.168.23.1 with username phone1/phone2 and proper secret, all should register and you should be good to go. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ott Rose *Sent:* Friday, July 10, 2009 8:33 AM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] setting up phones Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. -- Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -- Windows Live™: Keep your life in sync. Check it out.http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Video Call
To clarify, you have users 100, 101, and 102 on server 1 and 200, 201, and 202 on server 2. 100 can VC 101 and 102, but not 200-202. 100 can make a voice call to 200-202. Have you checked your iax.conf to make sure all codecs are functional? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Sent: Friday, July 10, 2009 8:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Video Call Hi, I have 2 asterisk servers link via IAX. and i'm trying to do a video call. if 2 sip users are registered on the same server, the video works fine. but if 1 sip user is on server 1 and sip user 2 on sip server 2. there's no video at all. is it because call from sip server 1 goes to sip server 2 via IAX? hope my question is clear enough, thanks in advanced Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
My bad. Asterisk does not register with the phone. It can send out SIP headers to make the phones re-register. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Friday, July 10, 2009 8:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] setting up phones Asterisk registers with the phones? Obviously I have zero experience with these sets, but that is a new one. Thanks, Steve On Fri, Jul 10, 2009 at 9:40 AM, Danny Nicholas da...@debsinc.com wrote: Let's draw this out and let you fill in the blanks. Your asterisk server has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. Sip.conf should look this [phone1] type=peer context=phones host=dynamic fromuser=phone1 secret=secret1 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone1:secr...@foobar.com/phone1 defaultip=192.168.23.2 mailbox=1001 disallow=all allow=alaw [phone2] type=peer context=phones host=dynamic fromuser=phone2 secret=secret2 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone2:secr...@foobar.com/phone2 defaultip=192.168.23.3 mailbox=1002 disallow=all allow=alaw assuming your phones are set up to contact 192.168.23.1 with username phone1/phone2 and proper secret, all should register and you should be good to go. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 8:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. _ Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) _ Windows LiveT: Keep your life in sync. Check it out. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_mobile help.
Sasa Bobek wrote: Could not agree more. I had chan_mobile up and running with an older version of Trix, but never managed to recreate it with the latest versions. Other people I talked to even suggested that it was made on purpose. With elastix the only problem I had was the missing mobile.conf.example, but you can create one from the Trix instructions from scratch or download it from the SVN. I've got a spare machine I can play with that on, I wish I could get it working on the server machine though. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Fwd: confirm f1ab6c493110edited]
Your membership in the mailing list asterisk-users has been disabled due to excessive bounces The last bounce received from you was dated Anybody else seeing this? My mail server logs don't show any issues. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Fwd: confirm f1ab6c493110edited]
Doug Lytle wrote: Your membership in the mailing list asterisk-users has been disabled due to excessive bounces The last bounce received from you was dated Anybody else seeing this? My mail server logs don't show any issues. Doug I just did yes, Don't know why :) Dunc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Fwd: confirm f1ab6c493110edited]
On 10 Jul 2009, at 15:30, Doug Lytle wrote: Your membership in the mailing list asterisk-users has been disabled due to excessive bounces The last bounce received from you was dated Anybody else seeing this? My mail server logs don't show any issues. Yeap. Happens quite regularly.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Fwd: confirm f1ab6c493110edited]
Anybody else seeing this? My mail server logs don't show any issues. Digging a little further shows that ASSP blocked several 'Pharmacy' spams from the list. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Fwd: confirm f1ab6c493110edited]
Dunc wrote: Doug Lytle wrote: Your membership in the mailing list asterisk-users has been disabled due to excessive bounces The last bounce received from you was dated Anybody else seeing this? My mail server logs don't show any issues. Doug I just did yes, Don't know why :) Dunc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users It's because two spam emails advertising canadian pharmaceuticals came in in rapid succession. Any rational filter would have bounced them to the sender (the sender being the Asterisk List Server), and will therefore be subject to the 'too many bounces' rules on the list. Which, I assume, means more than 1. Or possibly only 1. Honestly, those rules are a little undocumented. I get those messages often and find myself re-enabling my list membership on a regular basis due to spam that hits the list, but that my server likes to refuse. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI failover to SIP trunk
Tzafrir Cohen wrote:
On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote:
You have a small typo:
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten = _.,2,Dial(SIP,Provider,${EXTEN})
exten = _.,1,Dial(Zap/g1/${EXTEN})
exten = _.,2,Dial(SIP/Provider/${EXTEN})
('/' instead of ',')
While this will work, be aware that there are circumstances where you
may end up calling the number twice, once through each provider. One
example is if the number you dial is busy, that progress will be passed
via the PRI to asterisk and the dialplan will continue to the next
priority. In this case, dialing the number again through the SIP
provider. To avoid this you will need to use some dialplan logic and
check the result of the DIALSTATUS variable. See this page for examples:
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
-Dave
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Re: [asterisk-users] Video Call
hi sir yes you're correct, voice call works from 100 to 200-202 but not video call. on my iax i simply added: videosupport=yes allow=h264 allow=h263 TIA Ron Danny Nicholas wrote: To clarify, you have users 100, 101, and 102 on server 1 and 200, 201, and 202 on server 2. 100 can VC 101 and 102, but not 200-202. 100 can make a voice call to 200-202. Have you checked your iax.conf to make sure all codecs are functional? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Sent: Friday, July 10, 2009 8:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Video Call Hi, I have 2 asterisk servers link via IAX. and i'm trying to do a video call. if 2 sip users are registered on the same server, the video works fine. but if 1 sip user is on server 1 and sip user 2 on sip server 2. there's no video at all. is it because call from sip server 1 goes to sip server 2 via IAX? hope my question is clear enough, thanks in advanced Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Educational institutions: Your Asteriskexperiences wanted!
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Todd Sent: Thursday, July 09, 2009 10:50 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Educational institutions: Your Asteriskexperiences wanted! Hello! I've been asked to get a show of hands for some analysts for users in Higher Ed - Universities, Colleges, or any other 2 or 4 year degree- granting institutions. If this fits you, please let me know your contact data and briefly how you're using Asterisk, and if you don't mind I can pass your contact data along (for consumption by humans only - this is not a mailing list or marketing list) to some analysts who might be interested in talking to you about your Open Source experience(s). Sound interesting? Mail me back and we can discuss in detail. Also, I've been asked by a community member to create an .edu mailing list. I'm all for this, and I've been swamped with other things to do and haven't gotten around to it yet. Would you be interested in such a list that is specifically for discussing implementation issues for such higher educational institutions? JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ I would be very interested in a .edu mailing list! Dwight Hawley Executive Director of Information Technologies Northern Seminary dhaw...@seminary.edu (630) 620-2129 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
Carrier is bandwidth.com we are running Asterisk 1.6.1.1 i ran sip set debug on from the CLI Once i did a module reload it started displaying all the debuging info. Here is some of the debug info --- (13 headers 0 lines) --- Scheduling destruction of SIP dialog '5fafc3b57e3928877141d12f58c9f...@127.0.0.2' in 32000 ms (Method: REGISTER) [Jul 10 06:53:39] NOTICE[5641]: chan_sip.c:16397 handle_response_register: Outbound Registration: Expiry for dynamic is 120 sec (Scheduling reregistration in 105 s) --- SIP read from UDP://127.0.0.1:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK613dcb99;received=127.0.0.1;rport=5060 From: sip:5...@dynamic;tag=as51c22cdd To: sip:5...@dynamic;tag=as51c22cdd Call-ID: 7e1e2c4c702c5b1619fef39612192...@127.0.0.2 CSeq: 117 REGISTER Server: Asterisk PBX 1.6.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Expires: 120 Contact: sip:5...@127.0.0.1;expires=120 Date: Fri, 10 Jul 2009 10:53:39 GMT Content-Length: 0 Date: Fri, 10 Jul 2009 09:42:31 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Who is the carrier? What flavor of Asterisk are you using? Regardless, the phones should register and be able to call each other and other Asterisk apps if you have them in the dialplan. If you go to the Asterisk CLI and turn on SIP debugging, do you get anything at all? also, change registersip to yes. Thanks, Steve On Fri, Jul 10, 2009 at 9:32 AM, Ott Rose sixfourimp...@hotmail.com wrote: Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) Windows Live™: Keep your life in sync. Check it out. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) _ Insert movie times and more without leaving Hotmail®. http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutorial_QuickAdd_062009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
Extension 500 is registered just fine. 200 OK Maybe you should start with a GUI version of Asterisk. Try calling out via bandwidth with SIP verbose on and post your results. Call the other phone and post verbose. You do have logic in extensions.conf do you not? Thanks, Steve Totaro On Fri, Jul 10, 2009 at 10:58 AM, Ott Rose sixfourimp...@hotmail.comwrote: Carrier is bandwidth.com we are running Asterisk 1.6.1.1 i ran sip set debug on from the CLI Once i did a module reload it started displaying all the debuging info. Here is some of the debug info --- (13 headers 0 lines) --- Scheduling destruction of SIP dialog ' 5fafc3b57e3928877141d12f58c9f...@127.0.0.2' in 32000 ms (Method: REGISTER) [Jul 10 06:53:39] NOTICE[5641]: chan_sip.c:16397 handle_response_register: Outbound Registration: Expiry for dynamic is 120 sec (Scheduling reregistration in 105 s) --- SIP read from UDP://127.0.0.1:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060 ;branch=z9hG4bK613dcb99;received=127.0.0.1;rport=5060 From: sip:5...@dynamic;tag=as51c22cdd To: sip:5...@dynamic;tag=as51c22cdd Call-ID: 7e1e2c4c702c5b1619fef39612192...@127.0.0.2 CSeq: 117 REGISTER Server: Asterisk PBX 1.6.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Expires: 120 Contact: sip:5...@127.0.0.1 sip%3a...@127.0.0.1;expires=120 Date: Fri, 10 Jul 2009 10:53:39 GMT Content-Length: 0 -- Date: Fri, 10 Jul 2009 09:42:31 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Who is the carrier? What flavor of Asterisk are you using? Regardless, the phones should register and be able to call each other and other Asterisk apps if you have them in the dialplan. If you go to the Asterisk CLI and turn on SIP debugging, do you get anything at all? also, change registersip to yes. Thanks, Steve On Fri, Jul 10, 2009 at 9:32 AM, Ott Rose sixfourimp...@hotmail.comwrote: Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. -- Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.comwrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -- Windows Live™: Keep your life in sync. Check it out.http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI failover to SIP trunk
On Fri, Jul 10, 2009 at 10:45 AM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Tzafrir Cohen wrote:
On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote:
You have a small typo:
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten = _.,2,Dial(SIP,Provider,${EXTEN})
exten = _.,1,Dial(Zap/g1/${EXTEN})
exten = _.,2,Dial(SIP/Provider/${EXTEN})
('/' instead of ',')
While this will work, be aware that there are circumstances where you
may end up calling the number twice, once through each provider. One
example is if the number you dial is busy, that progress will be passed
via the PRI to asterisk and the dialplan will continue to the next
priority. In this case, dialing the number again through the SIP
provider. To avoid this you will need to use some dialplan logic and
check the result of the DIALSTATUS variable. See this page for examples:
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
-Dave
Good point.
I was unaware that busy back from a TDM circuit would progress in the
dialplan rather than going to the h exten.
What other cases are there like that?
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] [Fwd: confirm f1ab6c493110edited]
constantly but not over the last few days. one for each list. On Fri, Jul 10, 2009 at 10:30 AM, Doug Lytle supp...@drdos.info wrote: Your membership in the mailing list asterisk-users has been disabled due to excessive bounces The last bounce received from you was dated Anybody else seeing this? My mail server logs don't show any issues. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Lagged Extension
Hi There I have an extension which is in a different country and is constantly lagged (about 800ms). When anyone tries to call this extension we get a No route to destination message. Now I would have thought that the server should be able to find a route to the destination seeing as the peer poke finds it's way there. Or is that lag too much to create a SIP channel? Thanks in advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
so i filled in the info and now i get this when i run sip show peers Name/username HostDyn Nat ACL Port Status 500/500127.0.0.1D 5060 OK (1 ms) 501/501127.0.0.1D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] I still cannot call the extensions and the phones say no service on there screen From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 08:40:49 -0500 Subject: Re: [asterisk-users] setting up phones Let’s draw this out and let you fill in the blanks. Your asterisk server has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. Sip.conf should look this [phone1] type=peer context=phones host=dynamic fromuser=phone1 secret=secret1 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone1:secr...@foobar.com/phone1 defaultip=192.168.23.2 mailbox=1001 disallow=all allow=alaw [phone2] type=peer context=phones host=dynamic fromuser=phone2 secret=secret2 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone2:secr...@foobar.com/phone2 defaultip=192.168.23.3 mailbox=1002 disallow=all allow=alaw assuming your phones are set up to contact 192.168.23.1 with username phone1/phone2 and proper secret, all should register and you should be good to go. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 8:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) Windows Live™: Keep your life in sync. Check it out. _ Hotmail® has ever-growing storage! Don’t worry about storage limits. http://windowslive.com/Tutorial/Hotmail/Storage?ocid=TXT_TAGLM_WL_HM_Tutorial_Storage_062009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI failover to SIP trunk
Steve Totaro wrote:
On Fri, Jul 10, 2009 at 10:45 AM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Tzafrir Cohen wrote:
On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote:
You have a small typo:
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten = _.,2,Dial(SIP,Provider,${EXTEN})
exten = _.,1,Dial(Zap/g1/${EXTEN})
exten = _.,2,Dial(SIP/Provider/${EXTEN})
('/' instead of ',')
While this will work, be aware that there are circumstances where you
may end up calling the number twice, once through each provider. One
example is if the number you dial is busy, that progress will be passed
via the PRI to asterisk and the dialplan will continue to the next
priority. In this case, dialing the number again through the SIP
provider. To avoid this you will need to use some dialplan logic and
check the result of the DIALSTATUS variable. See this page for examples:
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
-Dave
Good point.
I was unaware that busy back from a TDM circuit would progress in the
dialplan rather than going to the h exten.
What other cases are there like that?
It is my understanding (through trial and error, reading, etc) that any
Dial command that does not result in an answered state will continue in
the dialplan after a timeout (if specified) or some sort of progress is
received. If the called channel results in an answer then dialplan
processing stops as soon as one party hangs up (unless the g option is
specified).
This works on any channels that can pass progress (SIP/IAX and Zap/DAHDI
PRI and BRI). Zap/DAHDI Analog channels are considered answered as soon
as the dial is complete so you won't be able to use this trick under
normal circumstances.
-Dave
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Re: [asterisk-users] setting up phones
I have the GUI setup and I setup users in the gui before. I still couldn't get it to work. I don't have any SIP trunks setup via the GUI because I can't figure out my settings and I was told I didn't need it to test extensions. I am not sure what you mean by Try calling out via bandwidth with SIP verbose on and post your results. Call the other phone and post verbose. You do have logic in extensions.conf do you not? I don't know how to do that. Date: Fri, 10 Jul 2009 11:08:59 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Extension 500 is registered just fine. 200 OK Maybe you should start with a GUI version of Asterisk. Try calling out via bandwidth with SIP verbose on and post your results. Call the other phone and post verbose. You do have logic in extensions.conf do you not? Thanks, Steve Totaro On Fri, Jul 10, 2009 at 10:58 AM, Ott Rose sixfourimp...@hotmail.com wrote: Carrier is bandwidth.com we are running Asterisk 1.6.1.1 i ran sip set debug on from the CLI Once i did a module reload it started displaying all the debuging info. Here is some of the debug info --- (13 headers 0 lines) --- Scheduling destruction of SIP dialog '5fafc3b57e3928877141d12f58c9f...@127.0.0.2' in 32000 ms (Method: REGISTER) [Jul 10 06:53:39] NOTICE[5641]: chan_sip.c:16397 handle_response_register: Outbound Registration: Expiry for dynamic is 120 sec (Scheduling reregistration in 105 s) --- SIP read from UDP://127.0.0.1:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK613dcb99;received=127.0.0.1;rport=5060 From: sip:5...@dynamic;tag=as51c22cdd To: sip:5...@dynamic;tag=as51c22cdd Call-ID: 7e1e2c4c702c5b1619fef39612192...@127.0.0.2 CSeq: 117 REGISTER Server: Asterisk PBX 1.6.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Expires: 120 Contact: sip:5...@127.0.0.1;expires=120 Date: Fri, 10 Jul 2009 10:53:39 GMT Content-Length: 0 Date: Fri, 10 Jul 2009 09:42:31 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Who is the carrier? What flavor of Asterisk are you using? Regardless, the phones should register and be able to call each other and other Asterisk apps if you have them in the dialplan. If you go to the Asterisk CLI and turn on SIP debugging, do you get anything at all? also, change registersip to yes. Thanks, Steve On Fri, Jul 10, 2009 at 9:32 AM, Ott Rose sixfourimp...@hotmail.com wrote: Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) Windows Live™: Keep your life in sync. Check it out. _ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
You are running asterisk as a local service (127.0.0.1 is localhost). You need to use the address from ifconfig (192.168.X.X) in sip.conf (bindaddr). This will make asterisk where your phones can talk to it and register. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 10:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones so i filled in the info and now i get this when i run sip show peers Name/username HostDyn Nat ACL Port Status 500/500127.0.0.1D 5060 OK (1 ms) 501/501127.0.0.1D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] I still cannot call the extensions and the phones say no service on there screen _ From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 08:40:49 -0500 Subject: Re: [asterisk-users] setting up phones Let's draw this out and let you fill in the blanks. Your asterisk server has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. Sip.conf should look this [phone1] type=peer context=phones host=dynamic fromuser=phone1 secret=secret1 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone1:secr...@foobar.com/phone1 defaultip=192.168.23.2 mailbox=1001 disallow=all allow=alaw [phone2] type=peer context=phones host=dynamic fromuser=phone2 secret=secret2 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone2:secr...@foobar.com/phone2 defaultip=192.168.23.3 mailbox=1002 disallow=all allow=alaw assuming your phones are set up to contact 192.168.23.1 with username phone1/phone2 and proper secret, all should register and you should be good to go. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 8:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. _ Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) _ Windows LiveT: Keep your life in sync. Check http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009 it out. _ HotmailR has ever-growing storage! Don't worry about storage limits. Check it out. http://windowslive.com/Tutorial/Hotmail/Storage?ocid=TXT_TAGLM_WL_HM_Tutori al_Storage_062009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI failover to SIP trunk
On Fri, Jul 10, 2009 at 11:40 AM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Steve Totaro wrote:
On Fri, Jul 10, 2009 at 10:45 AM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Tzafrir Cohen wrote:
On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote:
You have a small typo:
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten = _.,2,Dial(SIP,Provider,${EXTEN})
exten = _.,1,Dial(Zap/g1/${EXTEN})
exten = _.,2,Dial(SIP/Provider/${EXTEN})
('/' instead of ',')
While this will work, be aware that there are circumstances where you
may end up calling the number twice, once through each provider. One
example is if the number you dial is busy, that progress will be passed
via the PRI to asterisk and the dialplan will continue to the next
priority. In this case, dialing the number again through the SIP
provider. To avoid this you will need to use some dialplan logic and
check the result of the DIALSTATUS variable. See this page for examples:
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
-Dave
Good point.
I was unaware that busy back from a TDM circuit would progress in the
dialplan rather than going to the h exten.
What other cases are there like that?
It is my understanding (through trial and error, reading, etc) that any
Dial command that does not result in an answered state will continue in
the dialplan after a timeout (if specified) or some sort of progress is
received. If the called channel results in an answer then dialplan
processing stops as soon as one party hangs up (unless the g option is
specified).
This works on any channels that can pass progress (SIP/IAX and Zap/DAHDI
PRI and BRI). Zap/DAHDI Analog channels are considered answered as soon
as the dial is complete so you won't be able to use this trick under
normal circumstances.
-Dave
True I guess except that if the call fails as the OP posted, because the PRI
is down, it should work then right?
Another thing. For outbound calls, I do not have a timeout. So the user
hangs up when they are ready, or when the other side hangs up or gets
congestion, which amounts to the h exten, or am I not correct.
Why have a timeout on outbound dialing (unless you are a dialer app?) It is
not like voicemail where you want it to ring for so many seconds and then
roll to VM.
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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[asterisk-users] Dell poweredge T100 TE420
Hi All I am getting a strange prblem while installing TE420 on a Dell Poweredge T100 machine. I get a TE4XXX: version Synchronisation error nad the machine hangs which needs a hard reboot . Its a new machine and if i install TE410P then installation is successful. Strangely enough i have a working TE420F on another Dell POweredge T100 machine at a different locaiton ... The DELL tech support says his PCI express slots are fine ..has anyone encountered this problem before ? I've tried with 3 different TE420 cards to rule out a defective card Thanks Sriram ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
I saw 127.0.0.2, never seen that before. Loopback that I have seen is 127.0.0.1. I always just bind to 0.0.0.0 since I have never really seen a point to binding to a specific IP. I guess if you are dual homed and don't want remote phones to work, but then you could just block that stuff in IPTables or whatever firewall. Thanks, Steve T BTW, what GUI? That was part of what I was asking when I said what flavor of Asterisk? On Fri, Jul 10, 2009 at 11:51 AM, Danny Nicholas da...@debsinc.com wrote: You are running asterisk as a local service (127.0.0.1 is localhost). You need to use the address from ifconfig (192.168.X.X) in sip.conf (bindaddr). This will make asterisk where your phones can “talk” to it and register. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ott Rose *Sent:* Friday, July 10, 2009 10:33 AM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] setting up phones so i filled in the info and now i get this when i run sip show peers Name/username HostDyn Nat ACL Port Status 500/500127.0.0.1D 5060 OK (1 ms) 501/501127.0.0.1D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] I still cannot call the extensions and the phones say no service on there screen -- From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 08:40:49 -0500 Subject: Re: [asterisk-users] setting up phones Let’s draw this out and let you fill in the blanks. Your asterisk server has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. Sip.conf should look this [phone1] type=peer context=phones host=dynamic fromuser=phone1 secret=secret1 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone1:secr...@foobar.com/phone1 defaultip=192.168.23.2 mailbox=1001 disallow=all allow=alaw [phone2] type=peer context=phones host=dynamic fromuser=phone2 secret=secret2 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone2:secr...@foobar.com/phone2 defaultip=192.168.23.3 mailbox=1002 disallow=all allow=alaw assuming your phones are set up to contact 192.168.23.1 with username phone1/phone2 and proper secret, all should register and you should be good to go. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ott Rose *Sent:* Friday, July 10, 2009 8:33 AM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] setting up phones Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. -- Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -- Windows Live™: Keep your life in sync. Check it out.http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009 -- Hotmail® has ever-growing storage! Don’t worry about storage limits. Check it
[asterisk-users] Dropped Call Problem -- Looking for ideas and a consultant.
Hello Everyone. We have: Asterisk 1.4.21.2 zaptel-1.4.11 libpri-1.4.5 Sangoma A101D Connected to a PRI Cicso 7960G phones (About 30 of them) We have a problem with dropped calls that has going on for a long time. We get up to 5 dropped calls on a bad day. They all seem to be incoming calls. I have a recording of what my users report a dropped call sounds like right before it drops http://www.stepawayfromthecomputer.com/drop.wav Please have a listen to the recording and tell me what you think it means I am looking for any ideas as to what I should do to track this down. I would love a lead on a good consultant who can help fix this. Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
Great i changed it to my ip here is the debug and sip show peers. phones still say no service i get a dial tone when i pick it up and a busy signal when i call the other extension. Name/username HostDyn Nat ACL Port Status 500/50010.0.0.52D 5060 OK (1 ms) 501/50110.0.0.52D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] --- (12 headers 0 lines) --- Really destroying SIP dialog '1e56b3345aabbe4373a46fb973476...@10.0.0.52' Method: OPTIONS linux-zswk*CLI --- SIP read from UDP://10.0.0.52:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.52:5060;branch=z9hG4bK1210f27b;received=10.0.0.52;rport=5060 From: asterisk sip:aster...@10.0.0.52;tag=as66b3ded8 To: sip:5...@10.0.0.52;tag=as66b3ded8 Call-ID: 56cf67f56e3cba6602965f6317656...@10.0.0.52 CSeq: 102 OPTIONS Server: Asterisk PBX 1.6.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: sip:aster...@10.0.0.52 Accept: application/sdp Content-Length: 0 From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 10:51:18 -0500 Subject: Re: [asterisk-users] setting up phones You are running asterisk as a local service (127.0.0.1 is localhost). You need to use the address from ifconfig (192.168.X.X) in sip.conf (bindaddr). This will make asterisk where your phones can “talk” to it and register. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 10:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones so i filled in the info and now i get this when i run sip show peers Name/username HostDyn Nat ACL Port Status 500/500 127.0.0.1 D 5060 OK (1 ms) 501/501 127.0.0.1 D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] I still cannot call the extensions and the phones say no service on there screen From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 08:40:49 -0500 Subject: Re: [asterisk-users] setting up phones Let’s draw this out and let you fill in the blanks. Your asterisk server has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. Sip.conf should look this [phone1] type=peer context=phones host=dynamic fromuser=phone1 secret=secret1 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone1:secr...@foobar.com/phone1 defaultip=192.168.23.2 mailbox=1001 disallow=all allow=alaw [phone2] type=peer context=phones host=dynamic fromuser=phone2 secret=secret2 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone2:secr...@foobar.com/phone2 defaultip=192.168.23.3 mailbox=1002 disallow=all allow=alaw assuming your phones are set up to contact 192.168.23.1 with username phone1/phone2 and proper secret, all should register and you should be good to go. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 8:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP
Re: [asterisk-users] PRI failover to SIP trunk
Steve Totaro wrote:
On Fri, Jul 10, 2009 at 11:40 AM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Steve Totaro wrote:
On Fri, Jul 10, 2009 at 10:45 AM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Tzafrir Cohen wrote:
On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote:
You have a small typo:
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten = _.,2,Dial(SIP,Provider,${EXTEN})
exten = _.,1,Dial(Zap/g1/${EXTEN})
exten = _.,2,Dial(SIP/Provider/${EXTEN})
('/' instead of ',')
While this will work, be aware that there are circumstances where you
may end up calling the number twice, once through each provider. One
example is if the number you dial is busy, that progress will be passed
via the PRI to asterisk and the dialplan will continue to the next
priority. In this case, dialing the number again through the SIP
provider. To avoid this you will need to use some dialplan logic and
check the result of the DIALSTATUS variable. See this page for examples:
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
-Dave
Good point.
I was unaware that busy back from a TDM circuit would progress in the
dialplan rather than going to the h exten.
What other cases are there like that?
It is my understanding (through trial and error, reading, etc) that any
Dial command that does not result in an answered state will continue in
the dialplan after a timeout (if specified) or some sort of progress is
received. If the called channel results in an answer then dialplan
processing stops as soon as one party hangs up (unless the g option is
specified).
This works on any channels that can pass progress (SIP/IAX and Zap/DAHDI
PRI and BRI). Zap/DAHDI Analog channels are considered answered as soon
as the dial is complete so you won't be able to use this trick under
normal circumstances.
-Dave
True I guess except that if the call fails as the OP posted, because the PRI
is down, it should work then right?
I believe so, I haven't tried it. I imagine DIALSTATUS would be either
CHANUNAVAIL or CONGESTION.
Another thing. For outbound calls, I do not have a timeout. So the user
hangs up when they are ready, or when the other side hangs up or gets
congestion, which amounts to the h exten, or am I not correct.
I can't answer to the use of the h exten, I've never used it.
Why have a timeout on outbound dialing (unless you are a dialer app?) It is
not like voicemail where you want it to ring for so many seconds and then
roll to VM.
You usually wouldn't use a timeout for outbound PSTN calls. I only
mentioned it to try to be as complete as possible.
-Dave
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Re: [asterisk-users] setting up phones
Asterisk GUI-version : SVN-branch-2.0-r4962 Date: Fri, 10 Jul 2009 11:57:38 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones I saw 127.0.0.2, never seen that before. Loopback that I have seen is 127.0.0.1. I always just bind to 0.0.0.0 since I have never really seen a point to binding to a specific IP. I guess if you are dual homed and don't want remote phones to work, but then you could just block that stuff in IPTables or whatever firewall. Thanks, Steve T BTW, what GUI? That was part of what I was asking when I said what flavor of Asterisk? On Fri, Jul 10, 2009 at 11:51 AM, Danny Nicholas da...@debsinc.com wrote: You are running asterisk as a local service (127.0.0.1 is localhost). You need to use the address from ifconfig (192.168.X.X) in sip.conf (bindaddr). This will make asterisk where your phones can “talk” to it and register. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 10:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones so i filled in the info and now i get this when i run sip show peers Name/username HostDyn Nat ACL Port Status 500/500 127.0.0.1 D 5060 OK (1 ms) 501/501 127.0.0.1 D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] I still cannot call the extensions and the phones say no service on there screen From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 08:40:49 -0500 Subject: Re: [asterisk-users] setting up phones Let’s draw this out and let you fill in the blanks. Your asterisk server has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. Sip.conf should look this [phone1] type=peer context=phones host=dynamic fromuser=phone1 secret=secret1 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone1:secr...@foobar.com/phone1 defaultip=192.168.23.2 mailbox=1001 disallow=all allow=alaw [phone2] type=peer context=phones host=dynamic fromuser=phone2 secret=secret2 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone2:secr...@foobar.com/phone2 defaultip=192.168.23.3 mailbox=1002 disallow=all allow=alaw assuming your phones are set up to contact 192.168.23.1 with username phone1/phone2 and proper secret, all should register and you should be good to go. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 8:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) Windows Live™: Keep your life in sync. Check it out. Hotmail® has ever-growing storage! Don’t worry about storage limits. Check it out.
Re: [asterisk-users] setting up phones
Change the address in sip.conf, not the phone. On Fri, Jul 10, 2009 at 12:04 PM, Ott Rose sixfourimp...@hotmail.comwrote: Great i changed it to my ip here is the debug and sip show peers. phones still say no service i get a dial tone when i pick it up and a busy signal when i call the other extension. Name/username HostDyn Nat ACL Port Status 500/50010.0.0.52D 5060 OK (1 ms) 501/50110.0.0.52D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] --- (12 headers 0 lines) --- Really destroying SIP dialog '1e56b3345aabbe4373a46fb973476...@10.0.0.52' Method: OPTIONS linux-zswk*CLI --- SIP read from UDP://10.0.0.52:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.52:5060 ;branch=z9hG4bK1210f27b;received=10.0.0.52;rport=5060 From: asterisk sip:aster...@10.0.0.52 sip%3aaster...@10.0.0.52 ;tag=as66b3ded8 To: sip:5...@10.0.0.52 sip%3a...@10.0.0.52;tag=as66b3ded8 Call-ID: 56cf67f56e3cba6602965f6317656...@10.0.0.52 CSeq: 102 OPTIONS Server: Asterisk PBX 1.6.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: sip:aster...@10.0.0.52 sip%3aaster...@10.0.0.52 Accept: application/sdp Content-Length: 0 -- From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 10:51:18 -0500 Subject: Re: [asterisk-users] setting up phones You are running asterisk as a local service (127.0.0.1 is localhost). You need to use the address from ifconfig (192.168.X.X) in sip.conf (bindaddr). This will make asterisk where your phones can “talk” to it and register. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ott Rose *Sent:* Friday, July 10, 2009 10:33 AM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] setting up phones so i filled in the info and now i get this when i run sip show peers Name/username HostDyn Nat ACL Port Status 500/500127.0.0.1D 5060 OK (1 ms) 501/501127.0.0.1D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] I still cannot call the extensions and the phones say no service on there screen -- From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 08:40:49 -0500 Subject: Re: [asterisk-users] setting up phones Let’s draw this out and let you fill in the blanks. Your asterisk server has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. Sip.conf should look this [phone1] type=peer context=phones host=dynamic fromuser=phone1 secret=secret1 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone1:secr...@foobar.com/phone1 defaultip=192.168.23.2 mailbox=1001 disallow=all allow=alaw [phone2] type=peer context=phones host=dynamic fromuser=phone2 secret=secret2 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone2:secr...@foobar.com/phone2 defaultip=192.168.23.3 mailbox=1002 disallow=all allow=alaw assuming your phones are set up to contact 192.168.23.1 with username phone1/phone2 and proper secret, all should register and you should be good to go. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ott Rose *Sent:* Friday, July 10, 2009 8:33 AM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] setting up phones Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. -- Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone
Re: [asterisk-users] Dropped Call Problem -- Looking for ideas and a consultant.
We had the same problem using a Digium T1 card. We switched the coding to from NI2 to 5ess and we haven't dropped a call since. You will have to check with your service provider to see if they do 5ess. Connor Spiess Network Specialist -Original Message- From: Mark Engelhardt [mailto:ma...@intuitiveengineering.com] Sent: Friday, July 10, 2009 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dropped Call Problem -- Looking for ideas and a consultant. Hello Everyone. We have: Asterisk 1.4.21.2 zaptel-1.4.11 libpri-1.4.5 Sangoma A101D Connected to a PRI Cicso 7960G phones (About 30 of them) We have a problem with dropped calls that has going on for a long time. We get up to 5 dropped calls on a bad day. They all seem to be incoming calls. I have a recording of what my users report a dropped call sounds like right before it drops http://www.stepawayfromthecomputer.com/drop.wav Please have a listen to the recording and tell me what you think it means I am looking for any ideas as to what I should do to track this down. I would love a lead on a good consultant who can help fix this. Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Mail was checked for spam by the Freeware Edition of No Spam Today! The Freeware Edition is free for personal and non-commercial use. You can remove this notice by purchasing a full license! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
Phone 1 has 500 in all of it's id's and connects to server 10.0.0.52? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 11:05 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Great i changed it to my ip here is the debug and sip show peers. phones still say no service i get a dial tone when i pick it up and a busy signal when i call the other extension. Name/username HostDyn Nat ACL Port Status 500/50010.0.0.52D 5060 OK (1 ms) 501/50110.0.0.52D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] --- (12 headers 0 lines) --- Really destroying SIP dialog '1e56b3345aabbe4373a46fb973476...@10.0.0.52' Method: OPTIONS linux-zswk*CLI --- SIP read from UDP://10.0.0.52:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.52:5060;branch=z9hG4bK1210f27b;received=10.0.0.52;rport=5060 From: asterisk sip:aster...@10.0.0.52;tag=as66b3ded8 To: sip:5...@10.0.0.52;tag=as66b3ded8 Call-ID: 56cf67f56e3cba6602965f6317656...@10.0.0.52 CSeq: 102 OPTIONS Server: Asterisk PBX 1.6.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: sip:aster...@10.0.0.52 Accept: application/sdp Content-Length: 0 _ From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 10:51:18 -0500 Subject: Re: [asterisk-users] setting up phones You are running asterisk as a local service (127.0.0.1 is localhost). You need to use the address from ifconfig (192.168.X.X) in sip.conf (bindaddr). This will make asterisk where your phones can talk to it and register. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 10:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones so i filled in the info and now i get this when i run sip show peers Name/username HostDyn Nat ACL Port Status 500/500127.0.0.1D 5060 OK (1 ms) 501/501127.0.0.1D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] I still cannot call the extensions and the phones say no service on there screen _ From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 08:40:49 -0500 Subject: Re: [asterisk-users] setting up phones Let's draw this out and let you fill in the blanks. Your asterisk server has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. Sip.conf should look this [phone1] type=peer context=phones host=dynamic fromuser=phone1 secret=secret1 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone1:secr...@foobar.com/phone1 defaultip=192.168.23.2 mailbox=1001 disallow=all allow=alaw [phone2] type=peer context=phones host=dynamic fromuser=phone2 secret=secret2 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone2:secr...@foobar.com/phone2 defaultip=192.168.23.3 mailbox=1002 disallow=all allow=alaw assuming your phones are set up to contact 192.168.23.1 with username phone1/phone2 and proper secret, all should register and you should be good to go. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 8:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. _ Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is
Re: [asterisk-users] setting up phones
yes From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 11:20:29 -0500 Subject: Re: [asterisk-users] setting up phones Phone 1 has 500 in all of it’s id’s and connects to server 10.0.0.52? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 11:05 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Great i changed it to my ip here is the debug and sip show peers. phones still say no service i get a dial tone when i pick it up and a busy signal when i call the other extension. Name/username HostDyn Nat ACL Port Status 500/500 10.0.0.52 D 5060 OK (1 ms) 501/501 10.0.0.52 D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] --- (12 headers 0 lines) --- Really destroying SIP dialog '1e56b3345aabbe4373a46fb973476...@10.0.0.52' Method: OPTIONS linux-zswk*CLI --- SIP read from UDP://10.0.0.52:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.52:5060;branch=z9hG4bK1210f27b;received=10.0.0.52;rport=5060 From: asterisk sip:aster...@10.0.0.52;tag=as66b3ded8 To: sip:5...@10.0.0.52;tag=as66b3ded8 Call-ID: 56cf67f56e3cba6602965f6317656...@10.0.0.52 CSeq: 102 OPTIONS Server: Asterisk PBX 1.6.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: sip:aster...@10.0.0.52 Accept: application/sdp Content-Length: 0 From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 10:51:18 -0500 Subject: Re: [asterisk-users] setting up phones You are running asterisk as a local service (127.0.0.1 is localhost). You need to use the address from ifconfig (192.168.X.X) in sip.conf (bindaddr). This will make asterisk where your phones can “talk” to it and register. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 10:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones so i filled in the info and now i get this when i run sip show peers Name/username HostDyn Nat ACL Port Status 500/500 127.0.0.1 D 5060 OK (1 ms) 501/501 127.0.0.1D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] I still cannot call the extensions and the phones say no service on there screen From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 08:40:49 -0500 Subject: Re: [asterisk-users] setting up phones Let’s draw this out and let you fill in the blanks. Your asterisk server has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. Sip.conf should look this [phone1] type=peer context=phones host=dynamic fromuser=phone1 secret=secret1 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone1:secr...@foobar.com/phone1 defaultip=192.168.23.2 mailbox=1001 disallow=all allow=alaw [phone2] type=peer context=phones host=dynamic fromuser=phone2 secret=secret2 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = phone2:secr...@foobar.com/phone2 defaultip=192.168.23.3 mailbox=1002 disallow=all allow=alaw assuming your phones are set up to contact 192.168.23.1 with username phone1/phone2 and proper secret, all should register and you should be good to go. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 8:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Here is my physical network. We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. the other nic in the Asterisk box is plugged into your lan switch the phones are plugged into the lan switch I can ping the phones from the Asterisk server. Date: Thu, 9 Jul 2009 17:42:43 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to
Re: [asterisk-users] setting up phones
Here is my conf files. sip.conf [general] context=default port=5060 ; UDP port for Asterisk bindaddr=10.0.0.52; If we want to specify only an IP (if a computer has three different IPs) 0.0.0.0 means any IP srvlookup=yes ; Enable DNS SRV server [500] type=peer context=phones host=dynamic fromuser=500 secret=500 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = 500:5...@10.0.0.52/500 defaultip=10.0.0.60 mailbox=1001 disallow=all allow=alaw [501] type=peer context=phones host=dynamic fromuser=501 secret=501 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = 501:5...@10.0.0.52/501 defaultip=10.0.0.46 mailbox=1002 disallow=all allow=alaw == users.conf [501] username = 501 transfer = yes mailbox = 501 call-limit = 100 type = peer fullname = 501 registersip = yes host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 501 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = 501 nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 disallow = all allow = ulaw,gsm macaddress = 00085d10927f autoprov = yes label = 501 linenumber = 1 LINEKEYS = 1 [500] username = 500 transfer = yes mailbox = 500 call-limit = 100 type = peer fullname = 500 registersip = yes host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 500 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = 500 nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 macaddress = 00085d1095aa autoprov = yes label = 500 linenumber = 1 LINEKEYS = 1 disallow = all allow = ulaw,gsm Date: Fri, 10 Jul 2009 12:16:50 -0400 From: stot...@first-notification.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Change the address in sip.conf, not the phone. On Fri, Jul 10, 2009 at 12:04 PM, Ott Rose sixfourimp...@hotmail.com wrote: Great i changed it to my ip here is the debug and sip show peers. phones still say no service i get a dial tone when i pick it up and a busy signal when i call the other extension. Name/username HostDyn Nat ACL Port Status 500/50010.0.0.52D 5060 OK (1 ms) 501/50110.0.0.52D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] --- (12 headers 0 lines) --- Really destroying SIP dialog '1e56b3345aabbe4373a46fb973476...@10.0.0.52' Method: OPTIONS linux-zswk*CLI --- SIP read from UDP://10.0.0.52:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.52:5060;branch=z9hG4bK1210f27b;received=10.0.0.52;rport=5060 From: asterisk sip:aster...@10.0.0.52;tag=as66b3ded8 To: sip:5...@10.0.0.52;tag=as66b3ded8 Call-ID: 56cf67f56e3cba6602965f6317656...@10.0.0.52 CSeq: 102 OPTIONS Server: Asterisk PBX 1.6.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: sip:aster...@10.0.0.52 Accept: application/sdp Content-Length: 0 From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 10:51:18 -0500 Subject: Re: [asterisk-users] setting up phones You are running asterisk as a local service (127.0.0.1 is localhost). You need to use the address from ifconfig (192.168.X.X) in sip.conf (bindaddr). This will make asterisk where your phones can “talk” to it and register. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 10:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones so i filled in the info and now i get this when i run sip show peers Name/username HostDyn Nat ACL Port Status 500/500 127.0.0.1 D 5060 OK (1 ms) 501/501 127.0.0.1 D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] I still cannot call the extensions and the phones say no service on there screen From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 08:40:49 -0500 Subject: Re: [asterisk-users] setting up phones Let’s draw this out and let you fill in the blanks. Your asterisk server has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip address of 192.168.23.2. phone 2 has ip address of 192.168.23.3. Sip.conf should
Re: [asterisk-users] setting up phones
Bind to 0.0.0.0 put your phones on DHCP if they are not already and reboot. reload asterisk. turn on sip debugging call 501 from 500 post debug info. i bet it rings. On Fri, Jul 10, 2009 at 12:38 PM, Ott Rose sixfourimp...@hotmail.comwrote: Here is my conf files. sip.conf [general] context=default port=5060 ; UDP port for Asterisk bindaddr=10.0.0.52; If we want to specify only an IP (if a computer has three different IPs) 0.0.0.0 means any IP srvlookup=yes ; Enable DNS SRV server [500] type=peer context=phones host=dynamic fromuser=500 secret=500 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = 500:5...@10.0.0.52/500 defaultip=10.0.0.60 mailbox=1001 disallow=all allow=alaw [501] type=peer context=phones host=dynamic fromuser=501 secret=501 canreinvite=no directrtpsetup=no call-limit=3 nat=no qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = 501:5...@10.0.0.52/501 defaultip=10.0.0.46 mailbox=1002 disallow=all allow=alaw == users.conf [501] username = 501 transfer = yes mailbox = 501 call-limit = 100 type = peer fullname = 501 registersip = yes host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 501 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = 501 nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 disallow = all allow = ulaw,gsm macaddress = 00085d10927f autoprov = yes label = 501 linenumber = 1 LINEKEYS = 1 [500] username = 500 transfer = yes mailbox = 500 call-limit = 100 type = peer fullname = 500 registersip = yes host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 500 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = 500 nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 macaddress = 00085d1095aa autoprov = yes label = 500 linenumber = 1 LINEKEYS = 1 disallow = all allow = ulaw,gsm -- Date: Fri, 10 Jul 2009 12:16:50 -0400 From: stot...@first-notification.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Change the address in sip.conf, not the phone. On Fri, Jul 10, 2009 at 12:04 PM, Ott Rose sixfourimp...@hotmail.comwrote: Great i changed it to my ip here is the debug and sip show peers. phones still say no service i get a dial tone when i pick it up and a busy signal when i call the other extension. Name/username HostDyn Nat ACL Port Status 500/50010.0.0.52D 5060 OK (1 ms) 501/50110.0.0.52D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] --- (12 headers 0 lines) --- Really destroying SIP dialog '1e56b3345aabbe4373a46fb973476...@10.0.0.52' Method: OPTIONS linux-zswk*CLI --- SIP read from UDP://10.0.0.52:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.52:5060 ;branch=z9hG4bK1210f27b;received=10.0.0.52;rport=5060 From: asterisk sip:aster...@10.0.0.52;tag=as66b3ded8 To: sip:5...@10.0.0.52;tag=as66b3ded8 Call-ID: 56cf67f56e3cba6602965f6317656...@10.0.0.52 CSeq: 102 OPTIONS Server: Asterisk PBX 1.6.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: sip:aster...@10.0.0.52 Accept: application/sdp Content-Length: 0 -- From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 10:51:18 -0500 Subject: Re: [asterisk-users] setting up phones You are running asterisk as a local service (127.0.0.1 is localhost). You need to use the address from ifconfig (192.168.X.X) in sip.conf (bindaddr). This will make asterisk where your phones can “talk” to it and register. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ott Rose *Sent:* Friday, July 10, 2009 10:33 AM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] setting up phones so i filled in the info and now i get this when i run sip show peers Name/username HostDyn Nat ACL Port Status 500/500127.0.0.1D 5060 OK (1 ms) 501/501127.0.0.1D 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] I still
Re: [asterisk-users] setting up phones
Debug info is going to help the most here. Nobody is really going to look at your configs. I would also turn off lookup because if DNS fails, Asterisk doesn't care for it much. Try to hard code your IPs. Thanks, Steve On Fri, Jul 10, 2009 at 12:42 PM, Steve Totaro stot...@totarotechnologies.com wrote: Bind to 0.0.0.0 put your phones on DHCP if they are not already and reboot. reload asterisk. turn on sip debugging call 501 from 500 post debug info. i bet it rings. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped Call Problem -- Looking for ideas and a consultant.
This is an age old Asterisk (and general telephony) problem. I can't blame it all on Asterisk. Never thought of the 5ess, filed in my memory bank as this is an age old problem. Too bad it happens with SIP providers and not just the little guys but XO for instance. I hear crackling. Cell phones drop all the time. On a bad day I get five dropped cell phone calls a day. Thanks, Steve Totaro On Fri, Jul 10, 2009 at 12:15 PM, Connor Spiess cspi...@idea-ma.com wrote: We had the same problem using a Digium T1 card. We switched the coding to from NI2 to 5ess and we haven't dropped a call since. You will have to check with your service provider to see if they do 5ess. Connor Spiess Network Specialist -Original Message- From: Mark Engelhardt [mailto:ma...@intuitiveengineering.com] Sent: Friday, July 10, 2009 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dropped Call Problem -- Looking for ideas and a consultant. Hello Everyone. We have: Asterisk 1.4.21.2 zaptel-1.4.11 libpri-1.4.5 Sangoma A101D Connected to a PRI Cicso 7960G phones (About 30 of them) We have a problem with dropped calls that has going on for a long time. We get up to 5 dropped calls on a bad day. They all seem to be incoming calls. I have a recording of what my users report a dropped call sounds like right before it drops http://www.stepawayfromthecomputer.com/drop.wav Please have a listen to the recording and tell me what you think it means I am looking for any ideas as to what I should do to track this down. I would love a lead on a good consultant who can help fix this. Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Mail was checked for spam by the Freeware Edition of No Spam Today! The Freeware Edition is free for personal and non-commercial use. You can remove this notice by purchasing a full license! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped Call Problem -- Looking for ideas and a consultant.
Conner, I contacted my telco and they report they have a: EWSD Siemens Central Office Which does not support 5ess Any other way around this? How did you determine changing to 5ess would fix your problem? Mark On Jul 10, 2009, at 12:15 PM, Connor Spiess wrote: We had the same problem using a Digium T1 card. We switched the coding to from NI2 to 5ess and we haven't dropped a call since. You will have to check with your service provider to see if they do 5ess. Connor Spiess Network Specialist -Original Message- From: Mark Engelhardt [mailto:ma...@intuitiveengineering.com] Sent: Friday, July 10, 2009 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dropped Call Problem -- Looking for ideas and a consultant. Hello Everyone. We have: Asterisk 1.4.21.2 zaptel-1.4.11 libpri-1.4.5 Sangoma A101D Connected to a PRI Cicso 7960G phones (About 30 of them) We have a problem with dropped calls that has going on for a long time. We get up to 5 dropped calls on a bad day. They all seem to be incoming calls. I have a recording of what my users report a dropped call sounds like right before it drops http://www.stepawayfromthecomputer.com/drop.wav Please have a listen to the recording and tell me what you think it means I am looking for any ideas as to what I should do to track this down. I would love a lead on a good consultant who can help fix this. Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Mail was checked for spam by the Freeware Edition of No Spam Today! The Freeware Edition is free for personal and non-commercial use. You can remove this notice by purchasing a full license! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme problem (talk detection/opt) in 1.6.1.1
On Fri, 10 Jul 2009, Jared Mauch wrote: I need the 'talking' information to better identify rogue people on bridges. I'm a 1.2 Luddite so I don't have all these fancy new features :) A different solution to a similar problem. I had problems with abusive callers in my conferences. I whipped up some dialplan and AGI mojo to let an admin mute and unmute individual callers to identify the culprit and then kick that caller. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped Call Problem -- Looking for ideas and a consultant.
Steve, Thanks for your thoughts. I am tearing out my last bit of hair on this one. We only use sip on our internal network to talk to the 7960s We are getting drops from no-cell phone hard wired phones too. Unfortunately There are too many drops for me to let this go. :( Mark On Jul 10, 2009, at 12:48 PM, Steve Totaro wrote: This is an age old Asterisk (and general telephony) problem. I can't blame it all on Asterisk. Never thought of the 5ess, filed in my memory bank as this is an age old problem. Too bad it happens with SIP providers and not just the little guys but XO for instance. I hear crackling. Cell phones drop all the time. On a bad day I get five dropped cell phone calls a day. Thanks, Steve Totaro On Fri, Jul 10, 2009 at 12:15 PM, Connor Spiess cspi...@idea- ma.com wrote: We had the same problem using a Digium T1 card. We switched the coding to from NI2 to 5ess and we haven't dropped a call since. You will have to check with your service provider to see if they do 5ess. Connor Spiess Network Specialist -Original Message- From: Mark Engelhardt [mailto:ma...@intuitiveengineering.com] Sent: Friday, July 10, 2009 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dropped Call Problem -- Looking for ideas and a consultant. Hello Everyone. We have: Asterisk 1.4.21.2 zaptel-1.4.11 libpri-1.4.5 Sangoma A101D Connected to a PRI Cicso 7960G phones (About 30 of them) We have a problem with dropped calls that has going on for a long time. We get up to 5 dropped calls on a bad day. They all seem to be incoming calls. I have a recording of what my users report a dropped call sounds like right before it drops http://www.stepawayfromthecomputer.com/drop.wav Please have a listen to the recording and tell me what you think it means I am looking for any ideas as to what I should do to track this down. I would love a lead on a good consultant who can help fix this. Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Mail was checked for spam by the Freeware Edition of No Spam Today! The Freeware Edition is free for personal and non-commercial use. You can remove this notice by purchasing a full license! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped Call Problem -- Looking for ideas and a consultant.
DSS? Ask them what other signaling they can support. I would escalate every day if I were you. It is the only way to get things done. Get it to the top and be mad, even if you are not. When fixed, PRAISE everyone from top to bottom. A level one tech will say Ah you are using Asterisk, we don't support that, or Sorry, our switch cannot do 5ESS (when it can but takes a bit of work). Thanks, Steve Totaro On Fri, Jul 10, 2009 at 2:24 PM, Mark Engelhardt ma...@intuitiveengineering.com wrote: Conner, I contacted my telco and they report they have a: EWSD Siemens Central Office Which does not support 5ess Any other way around this? How did you determine changing to 5ess would fix your problem? Mark On Jul 10, 2009, at 12:15 PM, Connor Spiess wrote: We had the same problem using a Digium T1 card. We switched the coding to from NI2 to 5ess and we haven't dropped a call since. You will have to check with your service provider to see if they do 5ess. Connor Spiess Network Specialist -Original Message- From: Mark Engelhardt [mailto:ma...@intuitiveengineering.com] Sent: Friday, July 10, 2009 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dropped Call Problem -- Looking for ideas and a consultant. Hello Everyone. We have: Asterisk 1.4.21.2 zaptel-1.4.11 libpri-1.4.5 Sangoma A101D Connected to a PRI Cicso 7960G phones (About 30 of them) We have a problem with dropped calls that has going on for a long time. We get up to 5 dropped calls on a bad day. They all seem to be incoming calls. I have a recording of what my users report a dropped call sounds like right before it drops http://www.stepawayfromthecomputer.com/drop.wav Please have a listen to the recording and tell me what you think it means I am looking for any ideas as to what I should do to track this down. I would love a lead on a good consultant who can help fix this. Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Mail was checked for spam by the Freeware Edition of No Spam Today! The Freeware Edition is free for personal and non-commercial use. You can remove this notice by purchasing a full license! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] beeping in headsets from queue callers
How do I turn off the beeps in the head sets when customers are waiting in the Queue? Darryl Williams Information Technology Manager Direct: (214) 231-7325 Cell: (469) 583-6992 Fax: (262) 953-1929 Email: dwilli...@adeprocessing.com image001.jpg___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] beeping in headsets from queue callers
Darryl Williams schrieb: How do I turn off the beeps in the head sets when customers are waiting in the Queue? ringinuse=no in queues.conf and/or disable call waiting I guess. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
I don't see my extensions in my extensions.conf file. I see a bunch of other stuff but nothing that looks like this exten = 500,500,Dial (SIP/500,20,tr) I am guessing there should be something in there. Date: Fri, 10 Jul 2009 12:44:56 -0400 From: stot...@totarotechnologies.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Debug info is going to help the most here. Nobody is really going to look at your configs. I would also turn off lookup because if DNS fails, Asterisk doesn't care for it much. Try to hard code your IPs. Thanks, Steve On Fri, Jul 10, 2009 at 12:42 PM, Steve Totaro stot...@totarotechnologies.com wrote: Bind to 0.0.0.0 put your phones on DHCP if they are not already and reboot. reload asterisk. turn on sip debugging call 501 from 500 post debug info. i bet it rings. _ Insert movie times and more without leaving Hotmail®. http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutorial_QuickAdd_062009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped Call Problem -- Looking for ideas and a consultant.
More info on my dropped call issue: Here is a report on a dropped call from today: Call Started echoing then cut out Stats From the 7960 Stats Screen: RxCnt:011853 TxCnt:010204 MaxJtr: 762 RxLost: So, now I am starting to suspect that I have this problem: http://www.trixbox.org/forums/trixbox-forums/open-discussion/trixbox-ce-2-4-2-cisco-7960-7940-receiving-audio-drop-during-c Any suggestions? Mark On Jul 10, 2009, at 2:45 PM, Steve Totaro wrote: DSS? Ask them what other signaling they can support. I would escalate every day if I were you. It is the only way to get things done. Get it to the top and be mad, even if you are not. When fixed, PRAISE everyone from top to bottom. A level one tech will say Ah you are using Asterisk, we don't support that, or Sorry, our switch cannot do 5ESS (when it can but takes a bit of work). Thanks, Steve Totaro On Fri, Jul 10, 2009 at 2:24 PM, Mark Engelhardt ma...@intuitiveengineering.com wrote: Conner, I contacted my telco and they report they have a: EWSD Siemens Central Office Which does not support 5ess Any other way around this? How did you determine changing to 5ess would fix your problem? Mark On Jul 10, 2009, at 12:15 PM, Connor Spiess wrote: We had the same problem using a Digium T1 card. We switched the coding to from NI2 to 5ess and we haven't dropped a call since. You will have to check with your service provider to see if they do 5ess. Connor Spiess Network Specialist -Original Message- From: Mark Engelhardt [mailto:ma...@intuitiveengineering.com] Sent: Friday, July 10, 2009 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dropped Call Problem -- Looking for ideas and a consultant. Hello Everyone. We have: Asterisk 1.4.21.2 zaptel-1.4.11 libpri-1.4.5 Sangoma A101D Connected to a PRI Cicso 7960G phones (About 30 of them) We have a problem with dropped calls that has going on for a long time. We get up to 5 dropped calls on a bad day. They all seem to be incoming calls. I have a recording of what my users report a dropped call sounds like right before it drops http://www.stepawayfromthecomputer.com/drop.wav Please have a listen to the recording and tell me what you think it means I am looking for any ideas as to what I should do to track this down. I would love a lead on a good consultant who can help fix this. Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Mail was checked for spam by the Freeware Edition of No Spam Today! The Freeware Edition is free for personal and non-commercial use. You can remove this notice by purchasing a full license! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] beeping in headsets from queue callers
Darryl Williams escribió: How do I turn off the beeps in the head sets when customers are waiting in the Queue? Look for the option announce in queues.conf. -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Originate (Executing a System Command)
I know I'm doing something simple and wrong, but I can't quite figure it out: Example (executing system command): Action: Originate Channel: Local/1...@dummy Application: System http://www.voip-info.org/wiki/view/Asterisk+cmd+System Data: /path/to/script I keep getting a Unable to request channel and am not sure what it is looking for in place of Local/1...@dummy. The script is an internal voice delivery to my agents (among other things I'd like to do) Thanks! PB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
For now, you need these two lines in your dialplan - exten = 500,1,Dial(SIP/500,20,m) - exten = 501,1,Dial(SIP/501,20,m) This should let you dial your 2 extensions and hear MOH until it picks up _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 2:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones I don't see my extensions in my extensions.conf file. I see a bunch of other stuff but nothing that looks like this exten = 500,500,Dial (SIP/500,20,tr) I am guessing there should be something in there. _ Date: Fri, 10 Jul 2009 12:44:56 -0400 From: stot...@totarotechnologies.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Debug info is going to help the most here. Nobody is really going to look at your configs. I would also turn off lookup because if DNS fails, Asterisk doesn't care for it much. Try to hard code your IPs. Thanks, Steve On Fri, Jul 10, 2009 at 12:42 PM, Steve Totaro stot...@totarotechnologies.com wrote: Bind to 0.0.0.0 put your phones on DHCP if they are not already and reboot. reload asterisk. turn on sip debugging call 501 from 500 post debug info. i bet it rings. _ Insert movie times and more without leaving HotmailR. See how. http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutor ial_QuickAdd_062009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Originate (Executing a System Command)
Why not just Local/1 (unless your server is actually named dummy)? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of J. G. Sent: Friday, July 10, 2009 2:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Originate (Executing a System Command) I know I'm doing something simple and wrong, but I can't quite figure it out: Example (executing system command): Action: Originate Channel: Local/1...@dummy Application: System http://www.voip-info.org/wiki/view/Asterisk+cmd+System Data: /path/to/script I keep getting a Unable to request channel and am not sure what it is looking for in place of Local/1...@dummy. The script is an internal voice delivery to my agents (among other things I'd like to do) Thanks! PB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Originate (Executing a System Command)
J. G. schrieb: I know I'm doing something simple and wrong, but I can't quite figure it out: Example (executing system command): Action: Originate Channel: Local/1...@dummy Application: System http://www.voip-info.org/wiki/view/Asterisk+cmd+System Data: /path/to/script I keep getting a Unable to request channel and am not sure what it is looking for in place of Local/1...@dummy. Danny Nicholas schrieb: Why not just Local/1 (unless your server is actually named dummy)? For Local/ channels @... specifies the context, not a peer/hostname. Syntax: Local/extens...@context[/n] Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
I don't use Asterisk-GUI but the general idea of a GUI is so you don't have to modify files by hand. You use the graphical user interface to generate the entries you need. If you are using a GUI then don't touch the files. Just download EVB (Easy Vox Box) and use the GUI. If you want to mess with the conf files then download the source and compile it for a vanilla, non-gui installation. Pick one or the other until you know what you are doing. Thanks, Steve Totaro On Fri, Jul 10, 2009 at 4:11 PM, Ott Rose sixfourimp...@hotmail.com wrote: added that and still doesn't work. Is there a setting that could be set that requires me to dial a # * or something before the extension number? Plus the phones say no service? Should I reset them to factory and see if they pick up the right extensions from Asterisk? -- From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 14:46:22 -0500 Subject: Re: [asterisk-users] setting up phones For now, you need these two lines in your dialplan - exten = 500,1,Dial(SIP/500,20,m) - exten = 501,1,Dial(SIP/501,20,m) This should let you dial your 2 extensions and hear MOH until it picks up -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ott Rose *Sent:* Friday, July 10, 2009 2:39 PM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] setting up phones I don't see my extensions in my extensions.conf file. I see a bunch of other stuff but nothing that looks like this exten = 500,500,Dial (SIP/500,20,tr) I am guessing there should be something in there. -- Date: Fri, 10 Jul 2009 12:44:56 -0400 From: stot...@totarotechnologies.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Debug info is going to help the most here. Nobody is really going to look at your configs. I would also turn off lookup because if DNS fails, Asterisk doesn't care for it much. Try to hard code your IPs. Thanks, Steve On Fri, Jul 10, 2009 at 12:42 PM, Steve Totaro stot...@totarotechnologies.com wrote: Bind to 0.0.0.0 put your phones on DHCP if they are not already and reboot. reload asterisk. turn on sip debugging call 501 from 500 post debug info. i bet it rings. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
added that and still doesn't work. Is there a setting that could be set that requires me to dial a # * or something before the extension number? Plus the phones say no service? Should I reset them to factory and see if they pick up the right extensions from Asterisk? From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 14:46:22 -0500 Subject: Re: [asterisk-users] setting up phones For now, you need these two lines in your dialplan - exten = 500,1,Dial(SIP/500,20,m) - exten = 501,1,Dial(SIP/501,20,m) This should let you dial your 2 extensions and hear MOH until it picks up From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 2:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones I don't see my extensions in my extensions.conf file. I see a bunch of other stuff but nothing that looks like this exten = 500,500,Dial (SIP/500,20,tr) I am guessing there should be something in there. Date: Fri, 10 Jul 2009 12:44:56 -0400 From: stot...@totarotechnologies.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Debug info is going to help the most here. Nobody is really going to look at your configs. I would also turn off lookup because if DNS fails, Asterisk doesn't care for it much. Try to hard code your IPs. Thanks, Steve On Fri, Jul 10, 2009 at 12:42 PM, Steve Totaro stot...@totarotechnologies.com wrote: Bind to 0.0.0.0 put your phones on DHCP if they are not already and reboot. reload asterisk. turn on sip debugging call 501 from 500 post debug info. i bet it rings. Insert movie times and more without leaving Hotmail®. See how. _ Insert movie times and more without leaving Hotmail®. http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutorial_QuickAdd_062009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Originate (Executing a System Command)
J. G. escribió: I know I'm doing something simple and wrong, but I can't quite figure it out: Example (executing system command): Action: Originate Channel: Local/1...@dummy Application: System http://www.voip-info.org/wiki/view/Asterisk+cmd+System Data: /path/to/script I keep getting a Unable to request channel and am not sure what it is looking for in place of Local/1...@dummy. The script is an internal voice delivery to my agents (among other things I'd like to do) Thanks! PB Do you have the dummy context with the 1 extension in your dialplan with something like this? Did you reload it and check it within the CLI? [dummy] exten = 1,1,Answer() exten = 1,n,Wait(2) exten = 1,n,Hangup() The application (your external command) won't be launched until the Origiante channel is answered. -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
I don't think the GUI is editing the conf files correctly. I am not sure I have configure things right. At this point i think i am going to start from scratch. Date: Fri, 10 Jul 2009 16:19:52 -0400 From: stot...@totarotechnologies.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones I don't use Asterisk-GUI but the general idea of a GUI is so you don't have to modify files by hand. You use the graphical user interface to generate the entries you need. If you are using a GUI then don't touch the files. Just download EVB (Easy Vox Box) and use the GUI. If you want to mess with the conf files then download the source and compile it for a vanilla, non-gui installation. Pick one or the other until you know what you are doing. Thanks, Steve Totaro On Fri, Jul 10, 2009 at 4:11 PM, Ott Rose sixfourimp...@hotmail.com wrote: added that and still doesn't work. Is there a setting that could be set that requires me to dial a # * or something before the extension number? Plus the phones say no service? Should I reset them to factory and see if they pick up the right extensions from Asterisk? From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jul 2009 14:46:22 -0500 Subject: Re: [asterisk-users] setting up phones For now, you need these two lines in your dialplan - exten = 500,1,Dial(SIP/500,20,m) - exten = 501,1,Dial(SIP/501,20,m) This should let you dial your 2 extensions and hear MOH until it picks up From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Friday, July 10, 2009 2:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones I don't see my extensions in my extensions.conf file. I see a bunch of other stuff but nothing that looks like this exten = 500,500,Dial (SIP/500,20,tr) I am guessing there should be something in there. Date: Fri, 10 Jul 2009 12:44:56 -0400 From: stot...@totarotechnologies.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Debug info is going to help the most here. Nobody is really going to look at your configs. I would also turn off lookup because if DNS fails, Asterisk doesn't care for it much. Try to hard code your IPs. Thanks, Steve On Fri, Jul 10, 2009 at 12:42 PM, Steve Totaro stot...@totarotechnologies.com wrote: Bind to 0.0.0.0 put your phones on DHCP if they are not already and reboot. reload asterisk. turn on sip debugging call 501 from 500 post debug info. i bet it rings. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) _ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up phones
On Fri, 10 Jul 2009, Ott Rose wrote: I don't think the GUI is editing the conf files correctly. I am not sure I have configure things right. At this point i think i am going to start from scratch. Yea! -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)
Sorry to bump my own message - but had a mail server problem so don't
know if I missed any replys :(
Ta
Wayne.
Wayne wrote:
Hi all,
I've just built a new installation of CentOS release 5.3 (Final) and
have installed both
http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.1.1.tar.gzAsterisk
1.6.1.1 and subsequently Asterisk 1.6.0.10 (thinking that I was maybe
trying to be too cutting edge) on a Dell PowerEdge sc440 server (nothing
complex - Pentium Dual core 2ghz - 1gb ram - 70gb sata hd).
The setup at this point is real simple with one Cisco 7960 phone
registering with Asterisk using Skinny.
I'm finding that simple things as pressing any of the buttons on the
phone is enough to cause Asterisk to randomly restart from a
segmentation fault.
I've tried this with 1.6.1.1 and, after recompiling and replacing, 1.6.0.10.
I followed
http://www.voip-info.org/wiki/view/CentOS+5.2+and+Asterisk+1.6.x+installation
as a basis for installation leaving out things I didnt want to set up
(odbc / web admin ).
The only thing that didn't seem to go too well was the setup Dahdi
(dahdi-linux-2.2.0.1). Although I can do a 'make' and 'make install',
'make config' didnt work and there are no etc/dahdi/ directory to change
any config files (as suggested by the guide). This may not be related
but just in case I thought I would mention it.
This is from the console after pressing the 'speaker' button a couple of
times.
/usr/sbin/safe_asterisk: line 146: 21513 Segmentation fault (core
dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS}
${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal EXITSTATUS-128.
Automatically restarting Asterisk.
If I don't use the phone, Asterisk will stay running.
I can dial the 1000 test extension along with the 500 inter-asterisk
test, these seem to work as expected as long as I dial the number and
hit 'dial' on the phone rather than selecting the line and trying to
dial each digit in turn. If I try that then at some random point (but
not always) Asterisk will fault.
The firmware version on the phone is 7.2 to which I've had this phone
and several others running off a 1.2 setup for years (using
chan_skinny?) but thought it time to update Asterisk.
Anyone have any pointers please on what to check next?
Thanks,
Wayne
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Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)
On Fri, Jul 10, 2009 at 7:06 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
On Fri, Jul 10, 2009 at 6:44 PM, Wayne wa...@planetwayne.com wrote:
Sorry to bump my own message - but had a mail server problem so don't
know if I missed any replys :(
Ta
Wayne.
Wayne wrote:
Hi all,
I've just built a new installation of CentOS release 5.3 (Final) and
have installed both
http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.1.1.tar.gz
Asterisk
1.6.1.1 and subsequently Asterisk 1.6.0.10 (thinking that I was maybe
trying to be too cutting edge) on a Dell PowerEdge sc440 server (nothing
complex - Pentium Dual core 2ghz - 1gb ram - 70gb sata hd).
The setup at this point is real simple with one Cisco 7960 phone
registering with Asterisk using Skinny.
I'm finding that simple things as pressing any of the buttons on the
phone is enough to cause Asterisk to randomly restart from a
segmentation fault.
I've tried this with 1.6.1.1 and, after recompiling and replacing,
1.6.0.10.
I followed
http://www.voip-info.org/wiki/view/CentOS+5.2+and+Asterisk+1.6.x+installation
as a basis for installation leaving out things I didnt want to set up
(odbc / web admin ).
The only thing that didn't seem to go too well was the setup Dahdi
(dahdi-linux-2.2.0.1). Although I can do a 'make' and 'make install',
'make config' didnt work and there are no etc/dahdi/ directory to change
any config files (as suggested by the guide). This may not be related
but just in case I thought I would mention it.
This is from the console after pressing the 'speaker' button a couple of
times.
/usr/sbin/safe_asterisk: line 146: 21513 Segmentation fault (core
dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS}
${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal EXITSTATUS-128.
Automatically restarting Asterisk.
If I don't use the phone, Asterisk will stay running.
I can dial the 1000 test extension along with the 500 inter-asterisk
test, these seem to work as expected as long as I dial the number and
hit 'dial' on the phone rather than selecting the line and trying to
dial each digit in turn. If I try that then at some random point (but
not always) Asterisk will fault.
The firmware version on the phone is 7.2 to which I've had this phone
and several others running off a 1.2 setup for years (using
chan_skinny?) but thought it time to update Asterisk.
Anyone have any pointers please on what to check next?
Thanks,
Wayne
If you are set on beta then read no further then the next line.
File a bug report with a core dump.
OK opinion time.
Your server is more than adequate.
For my tastes, you are beyond bleeding edge on the Asterisk front.
Simply my opinion but if this is going to be a real production server
or something you want to use reliably then I would suggest.
1.4.Latest Zaptel
1.4.19 Asterisk (if infact that is the last version that had chan_zap and
not DAHDI)
1.4.Current LibPRI
And convert those phones to SIP, forget chan_skinny.
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)
On Fri, Jul 10, 2009 at 6:44 PM, Wayne wa...@planetwayne.com wrote:
Sorry to bump my own message - but had a mail server problem so don't
know if I missed any replys :(
Ta
Wayne.
Wayne wrote:
Hi all,
I've just built a new installation of CentOS release 5.3 (Final) and
have installed both
http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.1.1.tar.gz
Asterisk
1.6.1.1 and subsequently Asterisk 1.6.0.10 (thinking that I was maybe
trying to be too cutting edge) on a Dell PowerEdge sc440 server (nothing
complex - Pentium Dual core 2ghz - 1gb ram - 70gb sata hd).
The setup at this point is real simple with one Cisco 7960 phone
registering with Asterisk using Skinny.
I'm finding that simple things as pressing any of the buttons on the
phone is enough to cause Asterisk to randomly restart from a
segmentation fault.
I've tried this with 1.6.1.1 and, after recompiling and replacing,
1.6.0.10.
I followed
http://www.voip-info.org/wiki/view/CentOS+5.2+and+Asterisk+1.6.x+installation
as a basis for installation leaving out things I didnt want to set up
(odbc / web admin ).
The only thing that didn't seem to go too well was the setup Dahdi
(dahdi-linux-2.2.0.1). Although I can do a 'make' and 'make install',
'make config' didnt work and there are no etc/dahdi/ directory to change
any config files (as suggested by the guide). This may not be related
but just in case I thought I would mention it.
This is from the console after pressing the 'speaker' button a couple of
times.
/usr/sbin/safe_asterisk: line 146: 21513 Segmentation fault (core
dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS}
${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal EXITSTATUS-128.
Automatically restarting Asterisk.
If I don't use the phone, Asterisk will stay running.
I can dial the 1000 test extension along with the 500 inter-asterisk
test, these seem to work as expected as long as I dial the number and
hit 'dial' on the phone rather than selecting the line and trying to
dial each digit in turn. If I try that then at some random point (but
not always) Asterisk will fault.
The firmware version on the phone is 7.2 to which I've had this phone
and several others running off a 1.2 setup for years (using
chan_skinny?) but thought it time to update Asterisk.
Anyone have any pointers please on what to check next?
Thanks,
Wayne
If you are set on beta then read no further then the next line.
File a bug report with a core dump.
OK opinion time.
Your server is more than adequate.
For my tastes, you are beyond bleeding edge on the Asterisk front.
Simply my opinion but if this is going to be a real production server
or something you want to use reliably then I would suggest.
1.4.Latest Zaptel
1.4.19 Asterisk (if infact that is the last version that had chan_zap and
not DAHDI)
1.4.Current LibPRI
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)
Steve Totaro wrote: If you are set on beta then read no further then the next line. File a bug report with a core dump. OK opinion time. Your server is more than adequate. For my tastes, you are beyond bleeding edge on the Asterisk front. Simply my opinion but if this is going to be a real production server or something you want to use reliably then I would suggest. 1.4.Latest Zaptel 1.4.19 Asterisk (if infact that is the last version that had chan_zap and not DAHDI) 1.4.Current LibPRI And convert those phones to SIP, forget chan_skinny. Hi Steve, Thanks for the pointers. I must admit - I was leaning towards 1.6 as this apparently has support for SIP over TCP (?). My end goal with this was to try and get Asterisk talking to Exchange 2007 servers unified messaging. As for chan_skinny - I'm currently using this on an existing 1.2 server although from what I've picked up from previous posts (going back a while) the inbuilt version is now quite stable and possibly better than the older 'chan_skinny' (which I think the development has stopped for now?). This is why I opted to use it for the new 1.6 server. Still open to an suggestions though :) Thanks Wayne. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)
On Fri, Jul 10, 2009 at 7:33 PM, Wayne wa...@planetwayne.com wrote: Steve Totaro wrote: If you are set on beta then read no further then the next line. File a bug report with a core dump. OK opinion time. Your server is more than adequate. For my tastes, you are beyond bleeding edge on the Asterisk front. Simply my opinion but if this is going to be a real production server or something you want to use reliably then I would suggest. 1.4.Latest Zaptel 1.4.19 Asterisk (if infact that is the last version that had chan_zap and not DAHDI) 1.4.Current LibPRI And convert those phones to SIP, forget chan_skinny. Hi Steve, Thanks for the pointers. I must admit - I was leaning towards 1.6 as this apparently has support for SIP over TCP (?). My end goal with this was to try and get Asterisk talking to Exchange 2007 servers unified messaging. As for chan_skinny - I'm currently using this on an existing 1.2 server although from what I've picked up from previous posts (going back a while) the inbuilt version is now quite stable and possibly better than the older 'chan_skinny' (which I think the development has stopped for now?). This is why I opted to use it for the new 1.6 server. Still open to an suggestions though :) Thanks Wayne. Second line. File a bug report. There are not nearly as many people on 1.6 as 1.2 or 1.4. I wish I had stats, but many people from the old skool never wanted to go past 1.2, myself included. 1.4 has proven itself stable in my book with zaptel, I don't mess with DAHDI. I still do 1.2 installs for core systems. 1.4 was for app_rpt with the URI (a radio repeater controller that is USB based). I think the people using 1.4 have been early adoptors or just started using asterisk at that version. They don't even know about Asterisk .3 that only supported Adtran equipment. 1.6.X is super beta and you are in the new frontier. Out of the people using 1.6.x, you may be the only person to try chan_skinny. You may be the first to find the 1.6.x chan_skinny bug, so start asterisk so it does a core dump and make a skinny call, then file a bug report. Hopefully things on bugtracker have changed and it gets attention and not just closed or ignored. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Suggestions for web based soft phones
For a while now I've been looking for a good web based soft phone solution, but so far no luck. A few solutions which I've tried, both Java based and Flash based, either don't work, or had bad sound quality. I need something which I could put on my productions server for my clients. Seems like good web based solutions are all paid ones, nobody is giving it for free. Any ideas, suggestions whom to go with? Thanks -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)
On Fri, Jul 10, 2009 at 4:33 PM, Wayne wa...@planetwayne.com wrote: Hi Steve, Thanks for the pointers. I must admit - I was leaning towards 1.6 as this apparently has support for SIP over TCP (?). My end goal with this was to try and get Asterisk talking to Exchange 2007 servers unified messaging. While I haven't used the SIP over TCP in production (yet), I find that the 1.6.1 series is stable for our environment. I don't know about using Exchange, as we are staying as far from unified messaging as possible (for political reasons of course...) I wouldn't install 1.0, so why go back to 1.2 or 1.4. Just more to learn and relearn. The important thing is to have a test environment to get all of the show stopping buts out. As for chan_skinny - I'm currently using this on an existing 1.2 server although from what I've picked up from previous posts (going back a while) the inbuilt version is now quite stable and possibly better than the older 'chan_skinny' (which I think the development has stopped for now?). This is why I opted to use it for the new 1.6 server. What is the main reason for staying with skinny on these phones? I have quite a few 7940/7960 converted to SIP that work great. Next week I will try and duplicate this behavior on my test system with skinny, but you should get a bug report filed with the core and important configurations. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
