Re: [asterisk-users] Cisco 7971 behind NAT
You need to enable SIP transformations on the firewall, the packets will have to be dynamically re-written to handle multiple Cisco phones of these models. Be sure 'nat=no' is set in sip.conf for the phones as well, or Asterisk will reply to the incorrect ports (source instead of the mangled contact header). In this case, you'll need to compile in the SIP connection tracking/NAT bits in the kernel, they should be able to mangle the packets appropriately. I have never tested this, as all my deployments have hardware firewalls with SIP support built-in. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luki Sent: Monday, November 16, 2009 20:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco 7971 behind NAT Hi all, does anyone have any luck using a Cisco 7971 (SIP) behind NAT with two different accounts on the same server (i.e. two different extensions)? I am using Cisco-CP7971G-GE/8.3.0 and asterisk V1.4.something. The phone sends SIP packets from a high-numbered UDP port but expects a reply on port 5060. Fine, I do some magic with iptables to rewrite the packets (which limits me to one phone at that location, unless I'm mistaken). Incoming calls work fine on both accounts, but outgoing calls work only from the most recently registered account (the order is random due to timing) since both appear to asterisk as IP:5060. An outgoing call from the other account is rejected with an authentication mismatch, which makes sense. Asterisk matches the most recently registered peer by IP/port and if the user name differs, it complains, even if the password is the same for both accounts. So, is this the worst SIP implementation ever in those Cisco 7971's or am I doing something very wrong here? Technically even without NAT this confusion would occur as both accounts use IP:5060 so Asterisk cannot tell them apart during the initial peer matching stage. Of course the source port the Cisco selects is different with every dialog, so that doesn't help either. Any input would be appreciated before I throw that phone out of the window. Thanks, Luki ___ -- 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] RTPAUDIOQOS
I add this line in our in/out contexts:
exten = h,1,Noop(QOS=${RTPAUDIOQOS})
Then grep for 'QOS' in asterisk-verbose (assuming you have verbose logging on).
I'm sure you could output it anwhere else as well with a system call/echo.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
cov...@ccs.covici.com
Sent: Thursday, November 12, 2009 06:19
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RTPAUDIOQOS
OK, how do you get such information -- at times it would be very useful
to know.
Darryl Dunkin ddun...@netos.net wrote:
Sorry to reply so late, I am months behind and catching up.
I have been inspecting this on my own systems, and the results are
inconsistent to say the least. I’ve been dumping these to the verbose logs
for some time and monitoring them, but I have not been able to determine why
the numbers are so far off. I am more concerned with the packets lost due to
priority queuing within our network.
Here is an example just today:
ssrc=583450581
themssrc=1093951555
lp=0
rxjitter=0.003219
rxcount=1100
txjitter=0.000275
txcount=1108
rlp=57702
rtt=0.036000
If the txcount is only 1108, how can the remote lost packet count be 57702?
Unless the call was nearly inaudible?
I did verify with this end user, and the call was just fine. Is this an issue
with the phone at the remote end misreporting?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mindaugas Kezys
Sent: Tuesday, September 22, 2009 01:01
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] RTPAUDIOQOS
Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA
Sent: 2009 m. rugsėjo 22 d. 09:28
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] RTPAUDIOQOS
hey all,
can any body know what this parameter stands for
i got RTPAUDIOQOS while i have SIP channels
but could not understand then what this parameter tell
ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.00;txcount=83;rlp=0;rtt=14818.715000
if any one know plese help me to or give any documentation link
regards
Dhaval
Alternatives:
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How do
you spend it?
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cov...@ccs.covici.com
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Re: [asterisk-users] Silent Dialing
Thanks, that is what I checked, there is nothing in there that would appear to do it. I wasn't sure if there were any hidden variables I could set beforehand. I'll try the MOH class as it might work. The ringback tones are indicating that an external system is being called, and we are trying to integrate these as seamlessly as possible. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ivan Stepaniuk Sent: Wednesday, November 11, 2009 04:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Silent Dialing 'core show application dial' should give you an idea of what to play around with... In a similar scenario, once I used the 'm' option, with a special moh class. The moh class had some soft ticking sound because the remote system was not correctly indicating ringing, and sometimes delayed the audio ringing tone too much. This sorts of comforts users with a something-is-going on feedback sound, without having a double tone sequence. I don't know if it's the right way, it worked for me. I actually prefer to have two ringback tones. Darryl Dunkin wrote: Is there a way to disable ringing while dialing? Example, external users come into our IVR, and if they dial certain IVR options, these are sent off to a remote server for call handling ( Dial(SIP/extens...@remoteserver) for example). It rings once, then the remote system picks up. I would like it to be more transparent to the 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] RTPAUDIOQOS
Sorry to reply so late, I am months behind and catching up. I have been inspecting this on my own systems, and the results are inconsistent to say the least. I’ve been dumping these to the verbose logs for some time and monitoring them, but I have not been able to determine why the numbers are so far off. I am more concerned with the packets lost due to priority queuing within our network. Here is an example just today: ssrc=583450581 themssrc=1093951555 lp=0 rxjitter=0.003219 rxcount=1100 txjitter=0.000275 txcount=1108 rlp=57702 rtt=0.036000 If the txcount is only 1108, how can the remote lost packet count be 57702? Unless the call was nearly inaudible? I did verify with this end user, and the call was just fine. Is this an issue with the phone at the remote end misreporting? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mindaugas Kezys Sent: Tuesday, September 22, 2009 01:01 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] RTPAUDIOQOS Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: 2009 m. rugsėjo 22 d. 09:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RTPAUDIOQOS hey all, can any body know what this parameter stands for i got RTPAUDIOQOS while i have SIP channels but could not understand then what this parameter tell ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.00;txcount=83;rlp=0;rtt=14818.715000 if any one know plese help me to or give any documentation link regards Dhaval ___ -- 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] Silent Dialing
Is there a way to disable ringing while dialing? Example, external users come into our IVR, and if they dial certain IVR options, these are sent off to a remote server for call handling ( Dial(SIP/extens...@remoteserver) for example). It rings once, then the remote system picks up. I would like it to be more transparent to the 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] Breaking news, but what happened? 11.000 channels on one server
After dial.
I have put this in my hangup context as:
exten = h,1,Noop(QOS=${RTPAUDIOQOS})
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaus
Darilion
Sent: Thursday, August 27, 2009 13:04
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Breaking news,but what happened? 11.000
channels on one server
John Todd wrote:
5) Any summary stats on RTP packet loss, etc? (from
CHANNEL(rtpqos,audio,all)) on channels?
I wonder how to retrieve those stats:
- after Dial()?
- during Dial()? (how?)
regards
klaus
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Re: [asterisk-users] ODP: Re: Polycom Stop Downloading Config
Then remove the FTP/HTTP server from the configuration. You'll want to configure this in the boot loader by pressing the 'setup' softkey immediately after it boots, while giving the 5 second count-down. Clear the server name from the server options there. Then additionally, make sure your DHCP server is not handing out a boot server as well, as this will cause it to do the same thing. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peder Sent: Wednesday, June 17, 2009 13:29 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] ODP: Re: Polycom Stop Downloading Config What happens if the http server is down? My point is that I don't want it to try and pull any config from a server. I just want it to use its local config. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jacek Blaschke Sent: Wednesday, June 17, 2009 3:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] ODP: Re: Polycom Stop Downloading Config Go to the phone keyboard [menu] [settings] [advanced] [4-5-6] [admin settings] [network conf.] [sever menu]. Now (edit) change server type to http (or https) using left/right arrows. Save. Phone as all Polycom's will happily reboot and will not ask you again for tftp/ftp. You may have control over MOST of features from web interface (Polycom, 456). XML's are more powerful, but last saved config will remain to the next meeting with tftp. Some phones however will lost backgrounds downloaded from the server. Jacek - Wiadomość oryginalna - Od:: Peder pe...@networkoblivion.com Data:: środa, 17 Czerwiec 2009 21:43 Temat: Re: [asterisk-users] Polycom Stop Downloading Config But It still needs to hit the server to see that at some point. I just want it to stop pulling config totally, unless I tell it to. It is web based, so I would think there should be some way to only config it from the web interface, but I can't get it to stop tftp/ftp. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, June 17, 2009 10:46 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Polycom Stop Downloading Config Touch the syncinfo.xml file with a future time. This should tell the phone to stop polling. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peder Sent: Wednesday, June 17, 2009 10:27 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Polycom Stop Downloading Config Does anybody know of a way to tell the Polycom phones to stop trying to download their config? We have some setup for tftp and some for ftp and if they cannot reach the server, they just keep rebooting over and over and over and never stop. I would think it should try once or twice and stop, but it doesn't. ___ -- 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/listinf ___ -- 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/listinf ___ -- 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/listinf ___ -- 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] What causes this error?
Do you have an example of your configuration? I haven't converted my gateways to dahdi yet, but my configuration is, in this order: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson Sent: Wednesday, June 17, 2009 15:49 To: Asterisk User MailList Subject: [asterisk-users] What causes this error? [2009-05-27 02:06:16.294] WARNING[6971] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [2009-05-27 02:06:16.295] VERBOSE[6971] logger.c: [2009-05-27 02:06:16.295] == Primary D-Channel on span 1 up [2009-05-27 02:06:16.301] ERROR[6971] chan_dahdi.c: !! Got a UA, but i'm in state 7 I noticed the above error many days after this at around 2AM. This morning starting at about 2AM I got an endless stream of these errors until I restarted Asterisk. [2009-06-17 02:18:05.503] ERROR[30465] chan_dahdi.c: No more room in scheduler [2009-06-17 02:18:05.504] ERROR[30465] chan_dahdi.c: Asked to delete sched id -1??? [2009-06-17 02:18:05.504] ERROR[30465] chan_dahdi.c: No more room in scheduler [2009-06-17 02:18:07.103] ERROR[30465] chan_dahdi.c: No more room in scheduler [2009-06-17 02:18:07.103] ERROR[30465] chan_dahdi.c: Asked to delete sched id -1??? There are no messages in the full log file before these line since 21:43 on 6/16/2009. I am running asterisk 1.6.0.9, dahdi-linux-2.1.0.4, dahdi-tools-2.1.0.2, libpri-1.4.10 and wanpipe-3.5.2. The PRI line is plugged in to a Sangoma A102de. Any hints would be appreciated. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ ___ -- 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] What causes this error?
hardhdlc is for a BRI, use dchan=24 instead to set the d-channel. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson Sent: Wednesday, June 17, 2009 16:04 To: Asterisk User MailList Subject: Re: [asterisk-users] What causes this error? /etc/dahdi/system.conf has this: loadzone=us defaultzone=us #Sangoma A102 port 1 [slot:4 bus:7 span:1] wanpipe1 span=1,0,0,esf,b8zs bchan=1-23 hardhdlc=24 /etc/wanpipe/wanpipe1.conf has this: [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 4 PCIBUS = 7 FE_MEDIA= T1 FE_LCODE= B8ZS FE_FRAME= ESF FE_LINE= 1 TE_CLOCK = NORMAL TE_REF_CLOCK= 0 TE_HIGHIMPEDANCE= NO LBO = 0DB FE_TXTRISTATE= NO MTU = 1500 UDPPORT = 9000 TTL= 255 IGNORE_FRONT_END = NO TDMV_SPAN= 1 TDMV_DCHAN= 24 TDMV_HW_DTMF= YES [w1g1] ACTIVE_CH= ALL TDMV_ECHO_OFF= NO TDMV_HWEC= YES /etc/asterisk/chan_dahdi.conf has this: [trunkgroups] [channels] language=en usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=no transfer=yes canpark=yes cancallforward=yes callreturn=no echocancel=yes ;echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate = no busydetect=yes usesmdi=no ;Sangoma A102 port 1 [slot:4 bus:7 span:1] wanpipe1 group=1 context= switchtype=national echocancel=no signalling=pri_cpe channel =1-23 -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ From: Darryl Dunkin ddun...@netos.net Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wed, 17 Jun 2009 15:56:31 -0700 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Conversation: [asterisk-users] What causes this error? Subject: Re: [asterisk-users] What causes this error? Do you have an example of your configuration? I haven't converted my gateways to dahdi yet, but my configuration is, in this order: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson Sent: Wednesday, June 17, 2009 15:49 To: Asterisk User MailList Subject: [asterisk-users] What causes this error? [2009-05-27 02:06:16.294] WARNING[6971] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [2009-05-27 02:06:16.295] VERBOSE[6971] logger.c: [2009-05-27 02:06:16.295] == Primary D-Channel on span 1 up [2009-05-27 02:06:16.301] ERROR[6971] chan_dahdi.c: !! Got a UA, but i'm in state 7 I noticed the above error many days after this at around 2AM. This morning starting at about 2AM I got an endless stream of these errors until I restarted Asterisk. [2009-06-17 02:18:05.503] ERROR[30465] chan_dahdi.c: No more room in scheduler [2009-06-17 02:18:05.504] ERROR[30465] chan_dahdi.c: Asked to delete sched id -1??? [2009-06-17 02:18:05.504] ERROR[30465] chan_dahdi.c: No more room in scheduler [2009-06-17 02:18:07.103] ERROR[30465] chan_dahdi.c: No more room in scheduler [2009-06-17 02:18:07.103] ERROR[30465] chan_dahdi.c: Asked to delete sched id -1??? There are no messages in the full log file before these line since 21:43 on 6/16/2009. I am running asterisk 1.6.0.9, dahdi-linux-2.1.0.4, dahdi-tools-2.1.0.2, libpri-1.4.10 and wanpipe-3.5.2. The PRI line is plugged in to a Sangoma A102de. Any hints would be appreciated. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ ___ -- 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 ___ -- 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] Goto not matching
What does your 'On-net' context look like? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha Sent: Thursday, May 14, 2009 14:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Goto not matching Hi all, I'm using asterisk in realtime...I have a specific scenario to jump from context to another context...The call will come from a gateway registered under Test context and this call should be sent to the On-net extension as Listed in the paste bin below: http://pastebin.com/d50b2ba42 The issue is that the call is matching the test context but as soon as it execute the GoTo tag I got the following error in the log: Executing Goto(SIP/gw-in..net-b7803718, On-net|028945551|1) -- Goto (On-net,028945551,1) [May 14 20:38:13] WARNING[8462]: pbx.c:2470 __ast_pbx_run: Channel 'SIP/gw-in..net-b7803718' sent into invalid extension '028945551' in context 'On-net', but no invalid handler It seems that the GoTo is not working well here...Can someone help me in that please? 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] top posting again [was: Re: CDR Design]
Everyone read this top down for your IVR wav file. Press 9 for the company directory Press 8 for the billing department Press 1 for technical support Press 0 for the operator Next let us know who calls into your PBX complaining that your menu is whacked. Now discussing PBX related issues, that is on topic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of SIP Sent: Wednesday, December 17, 2008 15:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] top posting again [was: Re: CDR Design] Steve Edwards wrote: Top posting. Bottom posting. Honestly, if you can't use an effing scrollbar, please tell me so I can take you out back and beat you to death with something heavy. The .5 seconds it takes to scroll from one end of a message to another is no excuse for spending 2 minutes writing a tirade about how you don't like to spend that extra .5 seconds. I swear. You people need to get up, walk away from the computer, go outside and realise that this level of egocentrism is incredibly unhealthy. 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] Country numbering plan resources
Most vendors provide you a complete list of all destinations, descriptions, and rates when you sign up. It seems like the lists are already out there when/if you need them. Some countries, mobile rates differ, so they provide a large Excel sheet of all possible destinations, descriptions, and costs to load into your billing software. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of SIP Sent: Sunday, December 14, 2008 18:33 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Country numbering plan resources Jeff LaCoursiere wrote: On Sun, 14 Dec 2008, Tzafrir Cohen wrote: Right. So for those of us who want to do simple things and avoid complicated stuff such as telephony in shoddy continent of North America, could you please provide data for your country? So far we have AU, IL and NZ. Not that I am trying to put down the project, but I am struggling to understand how this will be useful to anyone. What will you actually *do* with this information once it is compiled? j Step 1: Compile a list of country codes broken down into landline/mobile to the best of anyone's random guesswork. Step 2: ??? Step 3: Profit!!! 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 ___ -- 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] CLI and choice of messages
I log verbose to a file and tail it. tail -f /var/log/asterisk/asterisk-verbose | grep Noop From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: Friday, December 05, 2008 13:44 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] CLI and choice of messages You'd think they'd actually have something like this. But nope, they don't. Only for debug, but no verbose output filtering. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: December 5, 2008 11:00 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] CLI and choice of messages Is there a way, for debugging purpose, to have a level where only Noop() cmds are shown in the CLI but nothing else in the dialplan appears (except for errors and warnings or course)? Mike ___ -- 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] OT: Polycom Firmware available (by accident?)
Instead, they are likely releasing something newer and better. I believe they have always had SIP software for download, however, it is never the most recent. They only provide 'previous software' for end-users, if you want the latest, you still have to go to your vendor. http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip_ upgrade.html There are some direct download links to the previous versions here, which are often newer than what is listed on the product support page (such as the 501s, only list something around 2.1.2): http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton Sent: Tuesday, November 11, 2008 08:50 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: Polycom Firmware available (by accident?) Jared Smith wrote: On Tue, 2008-11-11 at 10:41 -0500, Dave Fullerton wrote: Anyone looking for firmware should get it now before it disappears. It's my understanding that this isn't a fluke, but that Polycom has indeed changed their policy and will no longer you to go through your reseller to get the latest and greatest firmware. That's awesome! I had wondered that but since I hadn't seen links for 3.1.0B or the new BootROM's it made me a little suspicious. Thanks for the clarification. -Dave ___ -- 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] OT: Disable Polycom 650 Forward Softkey
In your phone configuration file, for all lines: divert divert.fwd.1.enabled = 0 divert.fwd.2.enabled = 0 divert.fwd.3.enabled = 0 divert.fwd.4.enabled = 0 divert.fwd.5.enabled = 0 divert.fwd.6.enabled = 0 / The worst part is this is the same softkey as 'hangup', bad design Polycom! When the remote user hangs up first and you use the softkey to hangup as well, you accidently end up forwarding somewhere (users freak out and hit random keys). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Andersen Sent: Friday, October 24, 2008 13:12 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] OT: Disable Polycom 650 Forward Softkey I've got a problem that keeps popping up with my reception phone. It is a IP 650 and the receptionist - on three occassions - has accidentally hit the Forward softkey just before she enters the Page All keystrokes and then all future calls get routed as an overhead page. I will admit, the first time it happened, I was totally stumped. Why the heck did I have customers yelling Hello, Hello, can you hear me over every single Polycom in the building. In retrospect, it was pretty funny. However, now that it has happened three, count 'em, three times, I've got to figure out how to disable that softkey. I've looked through the sip.cfg file and can't seem to figure out what option would remove that softkey. Has anyone ever had to do this? What feature should I disable? TIA Bill ___ -- 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] Sonicwall potentially causing long ping timesto SIP phones
From my experience, Sonicwall is a nightmare with SIP if you do not have Enhanced OS. General rules I use: -Do not use SIP transformations (the VOIP tab), these cause random RTP issues, and once you start forwarding calls between users, all things go to heck. You are better off using NAT/qualify in your sip.conf. -Do not use SonicOS Standard (all new Sonicwalls should come with Enhanced now anyway) as there is no method to increase the timeout for UDP rules, this will never be added to this firmware -In SonicOS Enhanced, create inbound and outbound permit rules for all UDP traffic to your PBX (assuming it is on the WAN side), set the UDP timeout to 300 or more, this covers SIP and RTP, but you can be more specific if you prefer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Michaelson Sent: Friday, October 24, 2008 13:41 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sonicwall potentially causing long ping timesto SIP phones Kristian Kielhofner wrote: On 10/23/08, Bruce Komito [EMAIL PROTECTED] wrote: We've had LOTS of problems with Sonicwalls doing bad things to SIP and RTP connections. I've seen the delay thing, as well as the Sonicwall throwing away entries from the ARP table because of inactivity. I've also seen sporadic, intermittent problems with transfer from one phone to another. I have no doubt that a new, properly configured Sonicwall can be made to function properly in a VoIP environment, but we are not Sonicwall experts, nor are many of the purported experts. In every case where we've had problems with VoIP behind a Sonicwall, the problems ALL disappear when we put the phones on a LAN segment that does not pass through the Sonicwall. So, now that's our going in position. If it works, great, but if it doesn't, our solution is to take the Sonicwall out of the picture. My $.02 . Bruce Komito WPTI Telecom (775) 236-5815 I wouldn't single out SonicWalls when it comes to breaking SIP traffic. Most of the anything but simple PAT devices I've seen that implement any SIP specific fixups usually end up breaking something along the line. Unless the product is from a company where SIP is their core competency (like Ingate, or /maybe/ Cisco) it's best to stay away and/or disable the SIP specific fixups wherever possible. I'm looking forward to the day when SIP-TLS is the norm and these devices have no idea what kind of traffic is flowing through them! - I sympathize, especially since a client of mine is facing the same situation. A potential update to their configuration involves exactly what you (Kristian) suggest: layering TLS in-between. I've run SIP/RTP and IAX over openVPN without issue routinely. What worries me is that the problem is not related to SIP awareness, and that some erratic performance by the Sonicwall that is benign in most circumstances manifests as a quality issue when carrying media streams. Seems unlikely, but does anybody have any clarity on this? ___ -- 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] Cisco 7960 not always receiving incoming calls
Oh, you are using ip inspect as well. I have this setup on a few routers when using the FW feature set: ip inspect udp idle-time 900 -Original Message- From: Stephen Reese [mailto:[EMAIL PROTECTED] Sent: Saturday, October 18, 2008 14:41 To: Asterisk Users Mailing List - Non-Commercial Discussion; Darryl Dunkin Subject: Re: [asterisk-users] Cisco 7960 not always receiving incoming calls I tried increasing the value and even set it to never and added the qualify line but that did not help. Do I need to poke any holes in the firewall on the nat device for the udp traffic to stay persistent? I have included my routers configuration in case someone notices something I may need to make the connection work correctly. Also when I call the phone within the OK reachable time after the call disconnects the status immediately become UNREACHABLE. ns1*CLIsip show peers Name/username HostDyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.1165060 Unmonitored 101/10168.156.63.118D N 1038 UNREACHABLE 3 sip peers [Monitored: 0 online, 1 offline Unmonitored: 2 online, 0 offline] [Oct 18 16:55:09] NOTICE[21216]: chan_sip.c:15231 handle_response_peerpoke: Peer '101' is now Reachable. (217ms / 2000ms) ns1*CLI sip show peers Name/username HostDyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.1165060 Unmonitored 101/10168.156.63.118D N 1038 OK (217 ms) 3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 0 offline] [Oct 18 17:24:16] NOTICE[21216]: chan_sip.c:19339 sip_p oke_noanswer: Peer '101' is now UNREACHABLE! Last qualify: 134 CISCO CONF FOLLOWS: ! version 12.4 service timestamps debug datetime msec service timestamps log datetime service password-encryption ! hostname 3725router ! boot-start-marker boot system flash:/c3725-adventerprisek9-mz.124-21.bin boot-end-marker ! logging buffered 8192 debugging logging console informational enable secret 5 ! aaa new-model ! ! aaa authentication login default local aaa authentication ppp default local aaa authorization exec default local aaa authorization network default local ! aaa session-id common clock timezone EST -5 clock summer-time PCTime date Apr 6 2003 2:00 Oct 26 2003 2:00 network-clock-participate slot 1 network-clock-participate slot 2 no ip source-route ! ip traffic-export profile IDS-SNORT interface FastEthernet0/0 bidirectional mac-address 000c.2989.f93a ip cef ! ! no ip dhcp use vrf connected ip dhcp excluded-address 172.16.2.1 ip dhcp excluded-address 172.16.3.1 ! ip dhcp pool VLAN2clients network 172.16.2.0 255.255.255.0 default-router 172.16.2.1 dns-server 205.152.144.23 205.152.132.23 option 66 ip 172.16.2.10 option 150 ip 172.16.2.10 ! ip dhcp pool VLAN3clients network 172.16.3.0 255.255.255.0 default-router 172.16.3.1 dns-server 205.152.144.23 205.152.132.23 ! ! ip domain name neocipher.net ip name-server 205.152.144.23 ip name-server 205.152.132.23 ip inspect name SDM_LOW cuseeme ip inspect name SDM_LOW dns ip inspect name SDM_LOW ftp ip inspect name SDM_LOW h323 ip inspect name SDM_LOW https ip inspect name SDM_LOW icmp ip inspect name SDM_LOW netshow ip inspect name SDM_LOW rcmd ip inspect name SDM_LOW realaudio ip inspect name SDM_LOW rtsp ip inspect name SDM_LOW sqlnet ip inspect name SDM_LOW streamworks ip inspect name SDM_LOW tftp ip inspect name SDM_LOW tcp ip inspect name SDM_LOW udp ip inspect name SDM_LOW vdolive ip inspect name SDM_LOW imap ip inspect name SDM_LOW pop3 ip inspect name SDM_LOW esmtp ip auth-proxy max-nodata-conns 3 ip admission max-nodata-conns 3 ip ips sdf location flash://256MB.sdf ip ips notify SDEE ip ips name sdm_ips_rule vpdn enable ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! crypto pki trustpoint TP-self-signed-995375956 enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-995375956 revocation-check none rsakeypair TP-self-signed-995375956 ! ! crypto pki certificate chain TP-self-signed-995375956 certificate self-signed 01 quit username user privilege 15 secret 5 ! ! ip ssh authentication-retries 2 ! ! crypto isakmp policy 3 encr 3des authentication pre-share group 2 ! crypto isakmp policy 10 hash md5 authentication pre-share crypto isakmp key cisco address 10.0.0.2 no-xauth ! crypto isakmp client configuration group VPN-Users key dns 2 domain neocipher.net pool VPN_POOL acl 115 include-local-lan netmask 255.255.255.0 crypto isakmp profile IKE-PROFILE match identity group VPN-Users client authentication list default isakmp authorization list default client configuration address initiate client configuration address respond virtual-template 1 ! ! crypto ipsec transform-set ESP-3DES-SHA esp-3des esp-sha
Re: [asterisk-users] Cisco 7960 not always receiving incoming calls
It is likely a NAT timeout issue. When you call outbound, you
'reactivate' the SIP session in your NAT device, allowing calls to come
in until it expires (default on many devices is 60 seconds). You may
also receive inbound calls when the phone reregisters regularly. Try
'qualify=yes' in your phones section in sip.conf to send keepalives
(option packets in this case) every two seconds to the phone to keep it
from going idle. You can see the state of the phone from the console
with a 'sip show peers', if unreachable, your NAT device has killed the
NAT forward.
Should look like one of these:
xxx/xxx x.x.x.x D N 5060 OK (46 ms)
xxx/xxx x.x.x.x D N 5060 UNREACHABLE
As another troubleshooting step, you can telnet to the phone and have it
reregister with Asterisk manually (register line 1 1) to see if that
brings it back to life.
If qualify doesn't do it, see if you can increase UDP timeouts in your
firewall/NAT device.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen
Reese
Sent: Friday, October 17, 2008 17:04
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco 7960 not always receiving incoming
calls
On Wed, Oct 15, 2008 at 7:57 PM, Stephen Reese [EMAIL PROTECTED]
wrote:
I've searched around and found a few similar situations where the
phone will call out when using a Asterisk server but not receive
inbound calls. My issue is a little stranger. If I call out from the
phone then the phone will receive the next inbound call. The phone
will not receive another inbound call until a call out again from it
first. Any ideas?
I am using SIP and am using the latest phone image from Cisco to date.
I am also using a Cisco router at the gateway. Is there anything
special I should to to make this work? Note my soft phone does not
have any issues using the same dialing rules and extension
information. Here is some of my config stuff:
ns1*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
vitel-outbound/rsreese 64.2.142.22 5060
Unmonitored
vitel-inbound/rsreese 64.2.142.1165060
Unmonitored
101/10168.156.63.118D N 1038
Unmonitored
3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0
offline]
Inbound call in progress when the SIP Cisco phone doesn't ring
Verbosity is at least 5
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/rsreese-082a8358,
default,101,1) in new stack
-- Goto (default,101,1)
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/rsreese-082a8358,
SIP/101SIP/[EMAIL PROTECTED],30) in new stack
== Using SIP RTP CoS mark 5
-- Called 101
== Using SIP RTP CoS mark 5
-- Called [EMAIL PROTECTED]
-- SIP/vitel-outbound-08270130 is making progress passing it to
SIP/rsreese-082a8358
-- SIP/vitel-outbound-08270130 is ringing
== Spawn extension (default, 101, 1) exited non-zero on
'SIP/rsreese-082a8358'
Inbound call in progress when the SIP Cisco does ring after I first
make an outbound call
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/rsreese-082a8358,
default,101,1) in new stack
-- Goto (default,101,1)
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/rsreese-082a8358,
SIP/101SIP/[EMAIL PROTECTED],30) in new stack
== Using SIP RTP CoS mark 5
-- Called 101
== Using SIP RTP CoS mark 5
-- Called [EMAIL PROTECTED]
-- SIP/101-0825cab8 is ringing
-- SIP/vitel-outbound-08270130 is making progress passing it to
SIP/rsreese-082a8358
-- SIP/vitel-outbound-08270130 is ringing
== Spawn extension (default, 101, 1) exited non-zero on
'SIP/rsreese-082a8358'
Extensions.conf, which I don't think is relevent, I've changed it to
just a simple dial the sip phone and it still fails.
exten = 101,1,Dial(SIP/101SIP/[EMAIL PROTECTED],30)
exten = 101,n,GotoIf($[${DIALSTATUS} =
CHANUNAVAIL]?lbl_default_1:)
exten = 101,n,GotoIf($[${DIALSTATUS} = NOANSWER]?lbl_default_1:)
exten = 101,n(lbl_default_0),Hangup()
exten = 101,n(lbl_default_1),Dial(SIP/[EMAIL PROTECTED],30)
exten = 101,n,Goto(lbl_default_0)
Cisco phone stuff from a Cisco 7960:
SIPDefault.cnf
image_version: P0S3-08-9-00
proxy1_address: neocipher.net; Can be dotted IP or FQDN
proxy_register: 1
messages_uri: 100
phone_password: cisco ; Limited to 31 characters (Default - cisco)
sntp_server:10.10.10.1
time_zone: EST
dial_template: DIALPLAN
nat_enable: 1
nat_address: 172.16.2.1
nat_received_processing: 1
outbound_proxy_port: 5060
outbond_proxy: ns1.neocipher.net
SIP0112B9EAFF72.cnf
image_version: P0S3-08-9-00
# Line 1 Setup
line1_name: 101
line1_authname: 101
line1_shortname: Line 101
line1_password: test
line1_displayname: Stephen Reese; # Line 1 Display Name (Display
name to use for SIP messaging)
# Line 2 Setup
#line2_name:
Re: [asterisk-users] Cisco 7960 not always receiving incoming calls
Sorry, I missed the Cisco router bit.
As a last resort (if qualify doesn't help), you could enter this
(global) to increase the timeout on UDP translations:
ip nat translation udp-timeout 300 (or greater if you prefer)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darryl
Dunkin
Sent: Friday, October 17, 2008 17:28
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco 7960 not always receiving incoming
calls
It is likely a NAT timeout issue. When you call outbound, you
'reactivate' the SIP session in your NAT device, allowing calls to come
in until it expires (default on many devices is 60 seconds). You may
also receive inbound calls when the phone reregisters regularly. Try
'qualify=yes' in your phones section in sip.conf to send keepalives
(option packets in this case) every two seconds to the phone to keep it
from going idle. You can see the state of the phone from the console
with a 'sip show peers', if unreachable, your NAT device has killed the
NAT forward.
Should look like one of these:
xxx/xxx x.x.x.x D N 5060 OK (46 ms)
xxx/xxx x.x.x.x D N 5060 UNREACHABLE
As another troubleshooting step, you can telnet to the phone and have it
reregister with Asterisk manually (register line 1 1) to see if that
brings it back to life.
If qualify doesn't do it, see if you can increase UDP timeouts in your
firewall/NAT device.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen
Reese
Sent: Friday, October 17, 2008 17:04
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco 7960 not always receiving incoming
calls
On Wed, Oct 15, 2008 at 7:57 PM, Stephen Reese [EMAIL PROTECTED]
wrote:
I've searched around and found a few similar situations where the
phone will call out when using a Asterisk server but not receive
inbound calls. My issue is a little stranger. If I call out from the
phone then the phone will receive the next inbound call. The phone
will not receive another inbound call until a call out again from it
first. Any ideas?
I am using SIP and am using the latest phone image from Cisco to date.
I am also using a Cisco router at the gateway. Is there anything
special I should to to make this work? Note my soft phone does not
have any issues using the same dialing rules and extension
information. Here is some of my config stuff:
ns1*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
vitel-outbound/rsreese 64.2.142.22 5060
Unmonitored
vitel-inbound/rsreese 64.2.142.1165060
Unmonitored
101/10168.156.63.118D N 1038
Unmonitored
3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0
offline]
Inbound call in progress when the SIP Cisco phone doesn't ring
Verbosity is at least 5
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/rsreese-082a8358,
default,101,1) in new stack
-- Goto (default,101,1)
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/rsreese-082a8358,
SIP/101SIP/[EMAIL PROTECTED],30) in new stack
== Using SIP RTP CoS mark 5
-- Called 101
== Using SIP RTP CoS mark 5
-- Called [EMAIL PROTECTED]
-- SIP/vitel-outbound-08270130 is making progress passing it to
SIP/rsreese-082a8358
-- SIP/vitel-outbound-08270130 is ringing
== Spawn extension (default, 101, 1) exited non-zero on
'SIP/rsreese-082a8358'
Inbound call in progress when the SIP Cisco does ring after I first
make an outbound call
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/rsreese-082a8358,
default,101,1) in new stack
-- Goto (default,101,1)
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/rsreese-082a8358,
SIP/101SIP/[EMAIL PROTECTED],30) in new stack
== Using SIP RTP CoS mark 5
-- Called 101
== Using SIP RTP CoS mark 5
-- Called [EMAIL PROTECTED]
-- SIP/101-0825cab8 is ringing
-- SIP/vitel-outbound-08270130 is making progress passing it to
SIP/rsreese-082a8358
-- SIP/vitel-outbound-08270130 is ringing
== Spawn extension (default, 101, 1) exited non-zero on
'SIP/rsreese-082a8358'
Extensions.conf, which I don't think is relevent, I've changed it to
just a simple dial the sip phone and it still fails.
exten = 101,1,Dial(SIP/101SIP/[EMAIL PROTECTED],30)
exten = 101,n,GotoIf($[${DIALSTATUS} =
CHANUNAVAIL]?lbl_default_1:)
exten = 101,n,GotoIf($[${DIALSTATUS} = NOANSWER]?lbl_default_1:)
exten = 101,n(lbl_default_0),Hangup()
exten = 101,n(lbl_default_1),Dial(SIP/[EMAIL PROTECTED],30)
exten = 101,n,Goto(lbl_default_0)
Cisco phone stuff from a Cisco 7960:
SIPDefault.cnf
image_version: P0S3-08-9-00
proxy1_address: neocipher.net; Can be dotted IP or FQDN
proxy_register: 1
messages_uri: 100
phone_password: cisco ; Limited to 31 characters (Default - cisco
Re: [asterisk-users] Budge Tones pick up wrong calls
Setting 'nat=yes' in your sip.conf for each phone will fix this. When set, Asterisk will ignore the ports defined in the SIP packet (always 5060 with the internal NAT IP) and instead use the IP and port the packet arrived on post-NAT. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Douglas Franklin Sent: Tuesday, October 14, 2008 14:53 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Budge Tones pick up wrong calls This sounds like a likely source of the problem. I changed the ports on two of the phones and the problem seems to have gone away. Thank you, Trevor, and others who responded. --Paul Trevor Peirce wrote: I have seen this with Polycom phones. In my case the problem turned out to be because there were several phones behind NAT and the NAT router got a little confused. The only solution I could find was to have the phones use different ports - ie. 5060, 5061, 5062. When they all shared 5060 the NAT router was unable to keep track of where an incoming call should be routed to. -- Paul Douglas Franklin Computer Manager, Union Gospel Mission of Yakima, Washington Husband of Danette Father of Laurene, Miriam, Tycko, Timothy, Sarabeth, Marie, Dawnita, Anna Leah, Alexander, and Caleb ___ -- 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] sometimes extensions can't be called
Try setting 'qualify=yes' in the sip.conf for the users. This will send a SIP options packet every two to the phone to verify the remote NAT device is allowing traffic from both sources to the phone. Afterwards, you'll usually see this status from the servers, to verify the phone is reachable: 123/12364.23.49.5 D N 15103OK (44 ms) If one server is unable to reach the phone, the status will instead be 'UNREACHABLE'. If it is a NAT device with a stateful firewall, it will likely only open the port for one source IP, and not both servers. Issues like this are why I run in an active/standby setup as opposed to active/active. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Ramos Sent: Wednesday, July 23, 2008 03:40 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] sometimes extensions can't be called Hi, I think i notice the problem now, but unfortunately i don't know how to fix it. i'm using 118103 i dial 113102 i got this on asterisk server #1. [Jul 23 18:27:48] -- Called 118102 [Jul 23 18:27:49] -- SIP/118102-08237ef0 is ringing what i did is keep on dialing then hang up dial then hang up, until i notice that when i dialed it went to asterisk #2 on asterisk 2 i see this: [Jul 23 18:30:40] -- Called 118102 but no ringing, it seems like it's trying to look for it, could it be because 102 is registered only on asterisk #1? but if i execute sip show peers i can see 118102 on both servers. i also had the problem wherein after i dial 118102, it goes to asterisk #2 and cince there is no ring, i hang up my phone, then i dialed again this time i see: [Jul 23 18:32:47] ERROR[17368]: chan_sip.c:3269 update_call_counter: Call to peer '118102' rejected due to usage limit of 2 yup i did set the limit to 2 but there was no asnwer on 118102 and i hangup, why did i reached the limit? Thanks in advanced Regards nhadie --- On Wed, 7/23/08, Darryl Dunkin [EMAIL PROTECTED] wrote: From: Darryl Dunkin [EMAIL PROTECTED] Subject: RE: [asterisk-users] sometimes extensions can't be called To: [EMAIL PROTECTED], asterisk-users@lists.digium.com Date: Wednesday, July 23, 2008, 5:13 AM Are the users registered to both active servers? ‘sip show peers’ in the console should make this obvious. If users are to call each other, they both need to be registered to the same server, or their client needs to be configured to register to both. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Ramos Sent: Tuesday, July 22, 2008 21:52 To: asterisk-users@lists.digium.com Subject: [asterisk-users] sometimes extensions can't be called Hi All, I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime on both asterisk. users register via domain, i have that domain on round-robin. users can register and sometimes can call each other, but sometimes even if an extension is register and i tried calling it, i got this on the the cli: [Jul 23 12:44:52] WARNING[32259]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jul 23 12:44:52] == Everyone is busy/congested at this time (1:0/0/1) but xlite or ip phone shows the extension is registered. but asterisk says it's busy. phones are behind NAT and using stun server. sip keep-alive is enabled onxlite or ip phone. but it's just very inconsistent. i don't know where to look at to fix this. any idea? nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP Packets Going To Wrong IP Address
What does the call setup look like on this? You can either debug sip in the console or 'ngrep -s 1500 -T -W byline host 75.36.34.98' From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nicholas Blasgen Sent: Monday, July 21, 2008 16:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RTP Packets Going To Wrong IP Address I have a user behind a firewall who's had no issues in the past connecting though his firewall. He's registered just fine. But when he places a call, a large number of them have no audio on either side of the connection. No one can hear him, he can't hear anyone as well. After a lot of poking around (and changing many settings) I noticed that Asterisk is communicating the RTP packets to an internal IP address. My server has no internal IP address, only an external address, so it's not like we're trying to route this anywhere else. As can be seen below, I've already identified the host as being behind a firewall and therefor to not trust packets from it. Anyone have a suggestion? Name/username HostDyn Nat ACL Port Status Realtime jfabriquer/jfabriquer 75.36.34.98 D N 55266OK (145 ms) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Asterisk SVN-branch-1.4-r118365 -- Nicholas Blasgen [EMAIL PROTECTED] 408.497.9796 (c) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sometimes extensions can't be called
Are the users registered to both active servers? 'sip show peers' in the console should make this obvious. If users are to call each other, they both need to be registered to the same server, or their client needs to be configured to register to both. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Ramos Sent: Tuesday, July 22, 2008 21:52 To: asterisk-users@lists.digium.com Subject: [asterisk-users] sometimes extensions can't be called Hi All, I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime on both asterisk. users register via domain, i have that domain on round-robin. users can register and sometimes can call each other, but sometimes even if an extension is register and i tried calling it, i got this on the the cli: [Jul 23 12:44:52] WARNING[32259]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jul 23 12:44:52] == Everyone is busy/congested at this time (1:0/0/1) but xlite or ip phone shows the extension is registered. but asterisk says it's busy. phones are behind NAT and using stun server. sip keep-alive is enabled onxlite or ip phone. but it's just very inconsistent. i don't know where to look at to fix this. any idea? nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cdr-custom/Master.csv rotation
It's like asking for directions, and someone tells you to drive,
useless.
Here is what we do here:
Create /etc/logrotate.d/asterisk:
/var/log/asterisk/asterisk-verbose /var/log/asterisk/messages
/var/log/asterisk/debug /var/log/asterisk/queue_log {
daily
rotate 7
compress
missingok
notifempty
sharedscripts
postrotate
/usr/local/bin/log_rot_ast
endscript
}
/usr/local/bin/log_rot_ast contains:
#!/bin/sh
/usr/sbin/asterisk -rx 'logger reload' /dev/null 21
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Hamilton
Sent: Saturday, June 14, 2008 19:05
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] cdr-custom/Master.csv rotation
Gavin,
I really do appreciate your one-liner. But is there any more insight
into
this? I know I have to use Logrotate, but I have no idea how I can
actually
get it done.
I'm going to try and figure it out right now, but for the benefit of the
list and archives, it just might be good if solutions could be posted
here
too.
Thanks,
Mark.
PS: Remember, many people get their answers from mailing list archives.
So
we'd rather get them solved than getting the same question on the list 3
months later. :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gavin
Henry
Sent: June 13, 2008 4:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] cdr-custom/Master.csv rotation
2008/6/13 Mark Hamilton [EMAIL PROTECTED]:
Hi,
How can I rotate /var/log/asterisk/cdr-custom/Master.csv nightly by
date?
Logrotate on a *nix box.
--
http://www.suretecsystems.com/services/openldap/
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Re: [asterisk-users] Interoffice phone setup
Are they just a trunk? Or are they your full PBX? If they are the full PBX, they handle the dialplan for dialing between phones, so there is no way around this. You would instead have to have your own Asterisk box at the same location as your phones, and use them for trunking if this is what you wanted to do. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph L. Casale Sent: Monday, June 09, 2008 15:52 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Interoffice phone setup We had an outage from our ISP this afternoon that cut prevented us from connecting to our SIP provider (someone physically cut a line downstream). All our phones inside the office stopped working as well? Why is that, and how can I set this up so phones can still dial each other inside the office? Thanks! jlc ___ -- 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] Logitech DiNovo Mini keyboard with myth
Wrong list? Or can you dial into Asterisk to setup recording of a show? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical Support Sent: Friday, June 06, 2008 18:59 To: 'Asterisk Users List' Subject: Re: [asterisk-users] Logitech DiNovo Mini keyboard with myth I found the necessary keyboard codes and created a mapping in .Xmodmap, and then finally: /usr/bin/xmodmap $HOME/.Xmodmap Still, myth doesn't seem to care about the new keysnow what? How do I make myth map these new codes to myth actions? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical Support Sent: June 6, 2008 9:03 PM To: Asterisk Users List Subject: [asterisk-users] Logitech DiNovo Mini keyboard with myth Has anyone create the necessary config/kbd file to allow the DiNovo mini to work well with myth? (Mapped all of the multimedia buttons etc) =MD= ___ -- 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] Zap Codec
Asterisk builds two channels and bridges them together. If the codecs mis-match then it must transcode, the negotiation on the Zap end is done seperately from the SIP end, so it does not care what your handset decided on. If you want ulaw, use ulaw, not g729 (on any call leg). You won't be able to mix and match codecs between calls, choose one for all calls and stick with it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann Sent: Tuesday, April 15, 2008 08:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec I guess that's my frustration, I don't want it g729, I want it ulaw, I just wish Zap did codec negotiation from the client. It'd be a nice option instead of automatically trying to translate if it's not ulaw. Could save some processor overhead(obviously at the expense of bandwidth). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, April 15, 2008 10:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec If you want to get a G729 call to go via Zap you must purchase a G729 license. No amount of discussion is going to change that. Jeremy Mann wrote: Sadly you are correct: -- Executing [EMAIL PROTECTED]:4] Set(SIP/156-083514c0, _SIP_CODEC=ulaw) in new stack -- Executing [EMAIL PROTECTED]:5] NoOp(SIP/156-083514c0, 4) in new stack -- Executing [EMAIL PROTECTED]:6] NoOp(SIP/156-083514c0, ) in new stack -- Executing [EMAIL PROTECTED]:7] Dial(SIP/156-083514c0, ) in new stack [Apr 15 09:51:25] WARNING[17434]: channel.c:3265 ast_request: No translator path exists for channel type Zap (native 76) to 256 [Apr 15 09:51:25] WARNING[17434]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'Zap' (cause 58 - Bearer capability not available) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:8] Hangup(SIP/156-083514c0, ) in new stack -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, April 15, 2008 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec That would work just spiffy if you are calling another SIP device, but by the time the call gets to that point in the dialplan the codec of the originating device has already been chosen and set in stone. Tilghman Lesher wrote: On Tuesday 15 April 2008 07:40:47 Jeremy Mann wrote: But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I only want ulaw used when SIPPEER-ZAP is the case. Set(_SIP_CODEC=ulaw) Dial(Zap/g0/...) -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are
Re: [asterisk-users] PBX Console
FOP works for us, no need for X: http://www.asternic.org If you need to avoid using a mouse, you can use the Polycom attendant console instead: http://www.polycom.com/usa/en/products/voice/desktop/soundpoint_ip/sound point_ip_attendant_console.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anonymous Sent: Tuesday, April 15, 2008 08:55 To: asterisk-users@lists.digium.com Subject: [asterisk-users] PBX Console Originally posted by: mailto: Hi, I've been looking into the one bad thing about * which is there's no practical solution to running a console. You know the kind where you have rows of buttons each representing an extension. You press the button of the extension you want to transfer the call to, and it's done. There's the beginnig of GUI version but it's going to eat resources for running X which can become less than desirable, besides it's not very competitive having to use a mouse to handle calls. Too slow. So my idea is to have a text window. We can run at a higher res than 25x80 and squeeze a fair number of extensions onto it. The idea is to either use the extension number to access an extension or for less than 100 station system, use a two digit number for each person. This way there's minimum typing for the operator. This have enough space to easily display busy, hold, vmail etc. as the status of each extension. This way with a flatscreen monitor, or dual for bigger systems we can even run the console away from the server and use minimum bandwidth. The other status screen would be a voice mail screen where you can A) see the status of voicemail. Lines in use etc. B) change the name and features associated with voice mail. -- Steve Szmidt HTML in e-mail is not safe. It let's spammers know to spam you more, and sets you up for online attack through IE 4.x and above. Using HTML in e-mail only promotes it as safe to the uninitiated. ___ Asterisk-Users mailing list mailto:Asterisk-Users@lists.digium.com 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] Zap Codec
Correct, those are two peers talking direct, one call leg (SIP-SIP). In this case, you have two call legs which are then bridged: SIP - Asterisk Asterisk - Zap You've already negotiated g729 before Asterisk notices that the call is going out Zap (via your dialplan). At this point, you have to transcode if your peer is set to use g729. Otherwise, force your SIP end to talk ulaw. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann Sent: Tuesday, April 15, 2008 11:45 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec Correct, but if I have two sip peers, one with G729ulaw, the other with gsmulaw, they will negotiate before trying to send audio. With ZAP, it tries to transcode whatever it receives into ulaw, period. No negotiation to even tell the client to send ulaw if capable. With no call level control(or dialplan logic, or anything!), I either use ulaw for ALL CALLS from sip peers(to other sip peers, to iax peers, to ZAP peers/channels), or use a combination of codecs and make sure it's able to be transcoded for the ZAP channels. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darryl Dunkin Sent: Tuesday, April 15, 2008 1:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec Asterisk builds two channels and bridges them together. If the codecs mis-match then it must transcode, the negotiation on the Zap end is done seperately from the SIP end, so it does not care what your handset decided on. If you want ulaw, use ulaw, not g729 (on any call leg). You won't be able to mix and match codecs between calls, choose one for all calls and stick with it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann Sent: Tuesday, April 15, 2008 08:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec I guess that's my frustration, I don't want it g729, I want it ulaw, I just wish Zap did codec negotiation from the client. It'd be a nice option instead of automatically trying to translate if it's not ulaw. Could save some processor overhead(obviously at the expense of bandwidth). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, April 15, 2008 10:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec If you want to get a G729 call to go via Zap you must purchase a G729 license. No amount of discussion is going to change that. Jeremy Mann wrote: Sadly you are correct: -- Executing [EMAIL PROTECTED]:4] Set(SIP/156-083514c0, _SIP_CODEC=ulaw) in new stack -- Executing [EMAIL PROTECTED]:5] NoOp(SIP/156-083514c0, 4) in new stack -- Executing [EMAIL PROTECTED]:6] NoOp(SIP/156-083514c0, ) in new stack -- Executing [EMAIL PROTECTED]:7] Dial(SIP/156-083514c0, ) in new stack [Apr 15 09:51:25] WARNING[17434]: channel.c:3265 ast_request: No translator path exists for channel type Zap (native 76) to 256 [Apr 15 09:51:25] WARNING[17434]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'Zap' (cause 58 - Bearer capability not available) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:8] Hangup(SIP/156-083514c0, ) in new stack -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, April 15, 2008 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec That would work just spiffy if you are calling another SIP device, but by the time the call gets to that point in the dialplan the codec of the originating device has already been chosen and set in stone. Tilghman Lesher wrote: On Tuesday 15 April 2008 07:40:47 Jeremy Mann wrote: But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I only want ulaw used when SIPPEER-ZAP is the case. Set(_SIP_CODEC=ulaw) Dial(Zap/g0/...) -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any
Re: [asterisk-users] Zap Codec
This is SIP channel you need to limit. Forcing ulaw only, the Polycom will fall back to ulaw. Per peer, in your sip.conf: disallow=all allow=ulaw From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann Sent: Monday, April 14, 2008 14:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Zap Codec Is there a way to force Zap channels to only use ulaw, and not even attempt g729 negotiation? My polycom 550 has G729,ulaw as priority, the Zap always fails b/c I'm not licensed for the codec on the asterisk box. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- 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] Pager (beeper) Emulation Script
I've done similar notifications in the dialplan.
It would probably look something like this:
exten = s,1,Read(PAGE,enter-phone-number10,10)
exten = s,2,System(/bin/echo Page content: ${PAGE} | /bin/mail -s
Page subject [EMAIL PROTECTED])
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andreas
van dem Helge
Sent: Thursday, February 21, 2008 21:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Pager (beeper) Emulation Script
Does anyone have a script that will emulate a normal numeric pager but
send the number to an email address? Also anyone happen to have the
traditional tones used in North America?
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Re: [asterisk-users] pulling my hair out over voicemail
Can you get some verbose output from your console/logs? It may be more obvious once you see what Asterisk is attempting to do when this extension is dialed. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Von Essen Sent: Wednesday, January 30, 2008 21:30 To: asterisk-users@lists.digium.com Subject: [asterisk-users] pulling my hair out over voicemail Ok, I have spent all night trying to figure this out, and hopefully somebody has a similar experience. I have a very basic asterisk config. Sample configs, with the only addition being by SIP phone, and my incoming voip. Last week I got everything setup, calls were working, etc.,. I cam across a tutorial for voicemail, followed it, and it worked. When I call my phone and dont answer, it goes to voicemail, and message is stored on server. I created an extension to retrieve the messages: exten = 1000,1,Ringing exten = 1000,2,Wait(2) exten = 1000,3,VoicemailMain And that worked. Granted, everything is still defaults, so when I dial 1000, I get the Comedian Mail greeting, then it prompts for mailbox and password, then I get the menu. Now, here is how it gets weird. Today I go and setup a new second SIP phone, and proceed to set it up for voicemail. Inbound calls go to voicemail properly when nobody answers, but I cant retrieve the messages. When I dial extension 1000, its rings for 2 seconds, then just goes silent. No greeting, no mailbox prompts, nothing. Any ideas what could be going on? I tried tweaking the extension 1000 so it looks like: exten = 1000,3,VoicemailMain,s6000 Where 6000 is my mailbox. But still nothing, when I dial 1000, it just goes silent. Please help. This is driving me nuts. I even tried re-installing asterisk from scratch - no change. -john ___ -- 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] pulling my hair out over voicemail
How about your sip.conf for your extensions? Example: [6001] host=dynamic type=friend disallow=all allow=ulaw I usually don't see this (I'm more production and haven't done heavy debug for a long time): [Jan 30 21:26:50] DEBUG[7917] channel.c: Set channel SIP/6001-081de7a8 to write format gsm [Jan 30 21:26:50] DEBUG[7917] channel.c: Set channel SIP/6001-081de7a8 to write format ulaw Since it's within the same second, I'm not sure which is actually being set. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Von Essen Sent: Wednesday, January 30, 2008 22:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] pulling my hair out over voicemail Tried it, but no change. A few updates. Even though I dont hear anything, if I hit a keys on the phone and then hang up, message log says: [Jan 30 21:26:57] WARNING[7917] app_voicemail.c: Unable to read password I enabled logging of everything, and the below is the snippet for when my SIP/6001 phone dial extension 1000 for Voicemail: [Jan 30 21:26:35] DEBUG[7917] pbx.c: Launching 'Ringing' [Jan 30 21:26:35] VERBOSE[7917] logger.c: -- Executing [EMAIL PROTECTED]:1] Ringing(SIP/6001-081de7a8, ) in new stack [Jan 30 21:26:35] DEBUG[7917] pbx.c: Launching 'Wait' [Jan 30 21:26:35] VERBOSE[7917] logger.c: -- Executing [EMAIL PROTECTED]:2] Wait(SIP/6001-081de7a8, 2) in new stack [Jan 30 21:26:37] DEBUG[7917] pbx.c: Launching 'VoiceMailMain' [Jan 30 21:26:37] VERBOSE[7917] logger.c: -- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/6001-081de7a8, [EMAIL PROTECTED]) in new stack [Jan 30 21:26:37] DEBUG[7917] app_voicemail.c: Before ast_answer [Jan 30 21:26:37] DEBUG[7917] devicestate.c: Notification of state change to be queued on device/channel SIP/6001-081de7a8 [Jan 30 21:26:37] DEBUG[7890] chan_sip.c: Checking device state for peer 6001 [Jan 30 21:26:37] DEBUG[7890] devicestate.c: Changing state for SIP/6001 - state 5 (Unavailable) [Jan 30 21:26:37] DEBUG[7890] chan_sip.c: Checking device state for peer 6001 [Jan 30 21:26:37] DEBUG[7917] chan_sip.c: SIP answering channel: SIP/6001-081de7a8 [Jan 30 21:26:37] DEBUG[7917] chan_sip.c: Setting framing from config on incoming call [Jan 30 21:26:37] DEBUG[7917] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True [Jan 30 21:26:37] DEBUG[7917] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jan 30 21:26:37] DEBUG[7917] chan_sip.c: -- Done with adding codecs to SDP [Jan 30 21:26:37] DEBUG[7917] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jan 30 21:26:37] DEBUG[7917] channel.c: Set channel SIP/6001-081de7a8 to write format gsm [Jan 30 21:26:37] DEBUG[7917] rtp.c: Ooh, format changed from unknown to ulaw [Jan 30 21:26:37] DEBUG[7917] rtp.c: Created smoother: format: 4 ms: 20 len: 160 [Jan 30 21:26:37] VERBOSE[7917] logger.c: -- SIP/6001-081de7a8 Playing 'vm-login' (language 'en') [Jan 30 21:26:37] DEBUG[7910] app_queue.c: Device 'SIP/6001' changed to state '5' (Unavailable) but we don't care because they're not a member of any queue. [Jan 30 21:26:37] DEBUG[7893] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Response 12349: Match Not Found [Jan 30 21:26:39] DEBUG[7917] channel.c: Set channel SIP/6001-081de7a8 to write format ulaw [Jan 30 21:26:50] DEBUG[7917] app_voicemail.c: Before find user for mailbox 8563682102 [Jan 30 21:26:50] DEBUG[7917] channel.c: Set channel SIP/6001-081de7a8 to write format gsm [Jan 30 21:26:50] DEBUG[7917] rtp.c: Difference is 82416, ms is 10322 [Jan 30 21:26:50] VERBOSE[7917] logger.c: -- SIP/6001-081de7a8 Playing 'vm-password' (language 'en') [Jan 30 21:26:50] DEBUG[7917] channel.c: Set channel SIP/6001-081de7a8 to write format ulaw [Jan 30 21:26:53] DEBUG[7893] chan_sip.c: Auto destroying SIP dialog '[EMAIL PROTECTED]' [Jan 30 21:26:53] DEBUG[7893] chan_sip.c: Destroying SIP dialog [EMAIL PROTECTED] [Jan 30 21:26:53] VERBOSE[7893] logger.c: Really destroying SIP dialog '[EMAIL PROTECTED]' Method: REGISTER [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: Sending dtmf: 53 (5), at 76.161.192.192 [Jan 30 21:26:55] DTMF[7917] channel.c: DTMF begin '5' received on SIP/6001-081de7a8 [Jan 30 21:26:55] DTMF[7917] channel.c: DTMF begin ignored '5' on SIP/6001-081de7a8 [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4)
Re: [asterisk-users] Call Parking with multiple lots
Look at app_valetparking, available here: http://www.freeswitch.org/asterisk_stuff/ I do not know about phone notification (I just use ringback/overhead paging), but it handles multiple contexts just fine. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron McCarthy Sent: Wednesday, January 23, 2008 15:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call Parking with multiple lots Hi List, I need to have one PBX but have multiple call parking for many different context. Basically for hosted VoIP, anyway this can be achineved? We really want to use the Snom's or something like that with a light on the phone so we can what caller is in each parking space/line. I have not seen anyway to do this, any ideals anyone? Thanks! ___ -- 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] Call Parking with multiple lots
I've had two live, it's a pretty archaic feature that emulates older PBXs so it isn't a popular feature at all. Just check the source on your options: -= Info about application 'ValetParkCall' =- [Synopsis] Valet Park Call [Description] ValetParkCall(exten|lotname|timeout[|return_ext][|return_pri][ |return_context]) Park Call at exten in lotname until someone calls ValetUnparkCall on the same exten + lotname set exten to 'auto' to auto-choose the slot. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron McCarthy Sent: Wednesday, January 23, 2008 16:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Parking with multiple lots How many contexts have you had this running on? And for the ring back, you cant have it park and then on the same call return the info, has to hangup then ring back? Thanks! On Jan 23, 2008 4:48 PM, Darryl Dunkin [EMAIL PROTECTED] wrote: Look at app_valetparking, available here: http://www.freeswitch.org/asterisk_stuff/ I do not know about phone notification (I just use ringback/overhead paging), but it handles multiple contexts just fine. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron McCarthy Sent: Wednesday, January 23, 2008 15:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call Parking with multiple lots Hi List, I need to have one PBX but have multiple call parking for many different context. Basically for hosted VoIP, anyway this can be achineved? We really want to use the Snom's or something like that with a light on the phone so we can what caller is in each parking space/line. I have not seen anyway to do this, any ideals anyone? Thanks! ___ -- 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] Rotating CDR records inside mysql - anyone does it?
You may speed up your queries with proper indexing. The default indexes are included with the table creation script here: http://www.voip-info.org/wiki-Asterisk+cdr+mysql ALTER TABLE `cdr` ADD INDEX ( `calldate` ); ALTER TABLE `cdr` ADD INDEX ( `dst` ); ALTER TABLE `cdr` ADD INDEX ( `accountcode` ); You could look at running a select/insert query to dump older CDRs off to an archive table (compressed, supports inserts and selects only): http://dev.mysql.com/tech-resources/articles/storage-engine.html After that's good, delete the older entires. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, January 22, 2008 11:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Rotating CDR records inside mysql - anyone does it? Hi everyone, I have a few asterisk machines doing PSTN calls, and I keep track of all cdr in a single machine running mysql 5. Since I have a very large amount of records in there, its getting pretty slow to query the database, so I'm wondering if anyone does some type of log rotating, like save the data for a single month inside a separate table and do that every month, so I keep the tables small enough to build my reports. I know this is mainly a mysql question, but maybe someone here has some stored procedures that do this already... Thanks for all help, Thiago Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! http://br.mail.yahoo.com/ ___ -- 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] Polycom Remotely Cancel Call Forward
In your per-phone configuration: phone1 reg ... divert divert.fwd.1.enabled = 0 divert.fwd.2.enabled = 0 divert.fwd.3.enabled = 0 divert.fwd.4.enabled = 0 divert.fwd.5.enabled = 0 divert.fwd.6.enabled = 0 / This removes the soft-key and disallows the option from the menu. I can't stand that feature as the soft-key is terribly misplaced, everytime you go hit 'end call', if the other user hangs up first, half our users ended up forwarding their phone to an invalid extension on accident. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Kiely Sent: Thursday, January 17, 2008 17:48 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Polycom Remotely Cancel Call Forward I guess I was interested in Disabling the forwarding feature completely via the config. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kai-Uwe Jensen Sent: Thursday, January 17, 2008 7:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom Remotely Cancel Call Forward When setting a forward on the phone, the phone will upload to your ftp server a modified macaddr-phone.cfg XML file that (amongst other locally made changes) contains an OVERRIDE statement similar to this: OVERRIDE reg.1.fwdContact= reg.1.fwdStatus=1 ... / Change the .fwdStatus attribute to 0, then reboot the phone (sip notify polycom-check-cfg peername). That will removed the forward just fine, at least in my setup here. Works the other way as well: modify the XML file to list a valid .fwdContact and set .fwdStatus to 1, then reboot the phone. That phone won't ring again until the forward is disabled :) ___ -- 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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.19.5/1228 - Release Date: 1/16/2008 9:01 AM ___ -- 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] Don't enter a queue if no one is logged in
Yes, in the queue config add: leavewhenempty = yes The users will enter the queue, but exit quickly and continue with the dialplan. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Pauly Sent: Sunday, December 09, 2007 12:33 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Don't enter a queue if no one is logged in I currently have the following setup: exten = 2000,1,Playback(/var/lib/asterisk/sounds/Greeting) exten = 2000,2,Queue(Qabcdef|t) exten = 2000,3,Playback(/var/lib/asterisk/sounds/EveryonesBusy) exten = 2000,4,Hangup exten = 2000,103,Hangup What happens is, that the greeting in step one is played regardless if anyone is logged into the queue. So immediately after the greet, we tell them we can't help them. What I would like is to check first to see if there is anyone logged into the queue, and then play the greeting. Is this possible? Is there a function that checks if anyone is logged in? ___ --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] SIP 7960 soft key customization?
I don't think you can do much with them. This is a good guide on the options you do have: http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/i pp7960/addprot/sip/admin/8_0/sipaxd75.htm -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Pauly Sent: Monday, December 10, 2007 07:06 To: asterisk-users@lists.digium.com Subject: [asterisk-users] SIP 7960 soft key customization? Does anyone know how to customize the order of the soft keys on a 7960 running SIP? All the documentation I could find is CallManager related. Specifically, I want to move the transfer function to the first set of buttons during a call. ___ --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 XML for configuration management, single-source-of-truth, etc.
You can store most of the configurations in a database which may be more accessable to you. Perl can also parse these configurations quickly enough if you know how to use the input record seperator ($/) properly. The only thing Asterisk will not store which you would probably need is the actual MAC address of the phones themselves. This may be done easily enough as comments in the users sip.conf section. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philip Prindeville Sent: Friday, December 07, 2007 13:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Using XML for configuration management,single-source-of-truth, etc. I'm starting work on some provisioning tools to simplify plugging in and configuring hard SIP handsets and conference bridges (maybe eventually MPEG-4 PoE video cameras that speak SIP as well). Issue is that I'd like to glean as much information out of the configuration files... but don't want to write a whole new parser to do it (especially not one that understands templates and macros). For instance, from the voicemail.conf, extensions.conf, and sip.conf files, I should be able to generate 90% of the configuration state needed for provisioning an out-of-the-box Sipura SPA941... if only those files were in some more parsable format, like XML. How much effort would it be to add an application that traverses the configuration state and writes it out as an XML flat file? Or perhaps at some point in the future, Asterisk's configuration files could be represented as XML natively (did someone in the back row just show gconf???). I'm a relative newbie, so if I'm missing something obvious or there's been a religious war on the subject in the past, apologies... -Philip ___ --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] G729 on wrong bus
Make sure you install the correct G729 module to match your platform, Digium provides both 32/64 versions. Then check to be sure your licenses are installed here properly: /var/lib/asterisk/licenses From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of broadband Voice Sent: Thursday, December 06, 2007 11:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] G729 on wrong bus when I do a show G729 i get a 0/0 even though I believe it is working for a carrier that accepts only g729. My feeling is becuase it is installed on 32 bus instead 64 bus thats why it is showing the wrong status. On 12/6/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: OkWhat is the issue? Does your G729 not work? Anyways who cares about the CPU? If you have a 32 bit Linux you need a 32 bit program. ___ --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] 7960 Won't Register Yet Multiple Attempts?
I've been struggling to get a stable NAT config on these forever :) Be sure the Netgear doesn't have a stateful firewall enabled (I believe 'SPI' is what they label the checkbox). These cheap boxes tend have flat 5 minute timeouts on UDP port translations and those kill the SIP port forward. The phone keeps sending new registration requests as it is not receiving the reply back through the NAT box. Not even setting qualify=yes will fix these sometimes (this should keep the port forward active and keep most NAT devices from timing out). With the Cisco, you can also telnet to the phone and force it to manually register on demand (ex: 'register 1 1') instead of rebooting, but usually useless once the NAT device flakes out. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, December 06, 2007 21:14 To: asterisk-users@lists.digium.com Subject: [asterisk-users] 7960 Won't Register Yet Multiple Attempts? Hi List, I've got a 7960 that's behind NAT (nat_enabled: 1 and nat_received_processing: 1) and for whatever reason doesn't seem to register, or at least hold a registration. If both the phone and the router (netgear) are rebooted, the phone will register, take a few incoming/outgoing calls no problems, then a few hours later, it drops the registration and never re-registers. If the phone itself is rebooted, I see a mess of registration attempts via SIP channels: 7X.183.246.XXX (None) 000e8XXX-5d 00101/00220 unkn No Rx: REGISTER 7X.183.246.XXX (None) 000e8XXX-5d 00101/00220 unkn No Rx: REGISTER 7X.183.246.XXX (None) 000e8XXX-5d 00101/00220 unkn No Rx: REGISTER 7X.183.246.XXX (None) 000e8XXX-5d 00101/00220 unkn No Rx: REGISTER 7X.183.246.XXX (None) 000e8XXX-5d 00101/00220 unkn No Rx: REGISTER 7X.183.246.XXX (None) 000e8XXX-5d 00101/00220 unkn No Rx: REGISTER Is there something that I'm missing. Short of replacing the customers router (which I have admin access to) is there anything else I should try? Any sort of packet filtering is disabling, nat is enabled in the SIP config, and port forwarding was also setup to forward 5060-5070 TCP and 1+ UDP to the phone to no avail. Note that if the phone is plugged directly into the customer's modem (thus removing the router out of the picture) the phone works perfectly. Thanks - Any input is appreciated -Robert Norton ___ --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] Asterisk server and DSCP QOS
We're using 184 here (aka TOS 5/EF). Not set by iptables though, instead it is set in sip.conf (tos_sip/tos_audio) and on our Polycom/Cisco phones. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Johnson Sent: Wednesday, December 05, 2007 12:49 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk server and DSCP QOS Can anyone comment on the DSCP quality of service settings on your Asterisk server? The network we're setting up has data on the default VLAN, Asterisk server and phones on VLAN 4, and we're using Polycom phones with a PC hooked up to the phone's pass-thru port. What iptables settings are you using on the Asterisk server for DSCP? What are your Polycom DSCP settings? We're using the Linksys SGE2000P POE switch which supports QOS via DSCP. Thanks a lot for any info. Steve ___ --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] G729/MOH Quality
Yes, it is in queues but there isn't anywhere to move them :) Instead we went ahead and generated whitenoise files just above the silence supression threshold to use as an alternate which is a little easier on the ears. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, November 30, 2007 16:59 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] G729/MOH Quality If the majority of the MoH is queues, move the location of the queue. ___ --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 server and DSCP QOS
Looks fine to me, you only need to specify DSCP or TOS (may need to check the manual for which it takes first). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Johnson Sent: Wednesday, December 05, 2007 14:02 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Asterisk server and DSCP QOS Thanks, Darryl, To clarify: in /etc/asterisk/sip.conf you have the lines: tos_sip=cs3; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. and in your Polycom configuration [I'm using Polycom's sip 2.2.0] you have something like (this is the one I'm uncertain about): QOS Ethernet RTP qos.ethernet.rtp.user_priority=5/ CallControl qos.ethernet.callControl.user_priority=5/ Other qos.ethernet.other.user_priority=2/ /Ethernet IP RTP qos.ip.rtp.dscp=184 qos.ip.rtp.min_delay=1 qos.ip.rtp.max_throughput=1 qos.ip.rtp.max_reliability=0 qos.ip.rtp.min_cost=0 qos.ip.rtp.precedence=5/ CallControl qos.ip.callControl.dscp=184 qos.ip.callControl.min_delay=1 qos.ip.callControl.max_throughput=0 qos.ip.callControl.max_reliability=0 qos.ip.callControl.min_cost=0 qos.ip.callControl.precedence=5/ /IP /QOS Thanks again! Steve Darryl Duncan wrote: We're using 184 here (aka TOS 5/EF). Not set by iptables though, instead it is set in sip.conf (tos_sip/tos_audio) and on our Polycom/Cisco phones. -Original Message- Subject: [asterisk-users] Asterisk server and DSCP QOS Can anyone comment on the DSCP quality of service settings on your Asterisk server? The network we're setting up has data on the default VLAN, Asterisk server and phones on VLAN 4, and we're using Polycom phones with a PC hooked up to the phone's pass-thru port. What iptables settings are you using on the Asterisk server for DSCP? What are your Polycom DSCP settings? We're using the Linksys SGE2000P POE switch which supports QOS via DSCP. Thanks a lot for any info. Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729/MOH Quality
Does anyone have any opinions on the music on hold quality over G729? The stock files seem to sound terrible over it, this is enhanced further by calls coming from the PSTN via a Zaptel gateway. I am only using the stock wav files and have not attempted to use much else so far. I've ruled out timing issues on the system generating the MOH itself (ztdummy on the PBX itself, our Zaptel gateway is a separate Asterisk server). There is no transcoding going on in the middle except via our Zaptel/T1 gateway. When using G711 it sounds fine of course, but this doesn't work well for remote sites with lower bandwidth connections. As of now, I'm torn between getting complaints from end users about the music or killing it entirely (leaving people waiting in queues with dead silence). ___ --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] Wanted: tutorial on troubleshooting SIP issues
For the non-GUI guys on the server end: ngrep -s 1500 -T -W byline host phone IP and udp port 5060 Add -O file to dump to a file for later Wireshark viewing on your local system. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alan Lord Sent: Friday, November 09, 2007 15:39 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Wanted: tutorial on troubleshooting SIP issues Steve Edwards wrote: snip / Examples of what I'd like to see: 1) A SIP telephone registering successfully. 2) A SIP telephone failing to register for reasons x, y, and z. snip / I'm sorry but I don't see this as being very hard. Just install Wireshark and do it yourself... Alan ___ --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 - how to protect the conference?
You could use meetme realtime and have the admin update the pin via a web interface instead. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ondrej Valousek Sent: Monday, November 05, 2007 09:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Meetme - how to protect the conference? Hi all, I am just wondering - it there any way how to protect a conference from being abused by someone? I know I can request pin, but that pin is then hardcoded in meetme.conf and normal user can not change it. I would like to establish an admin user who could set a pin for the conference to be used by other participants. Is that possible? Thanks, Ondrej ___ --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] Polycom and NAT
That should do it, it tells Asterisk to override the contact field which includes the private IP, and use the public IP and port it received the packet from instead. Try a 'sip debug peer peer' and see what it is coming in as. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C Sent: Wednesday, August 22, 2007 05:33 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom and NAT I have both of those command lines for my natted sip device. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darryl Dunkin Sent: Wednesday, 22 August 2007 1:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom and NAT In your sip.conf, for the user: nat=yes To send keepalives for the UDP connection (depending on how flimsy the device handling NAT is): qualify=yes From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C Sent: Tuesday, August 21, 2007 17:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom and NAT Hi All, I have a Polycom 501 that is behind a NAT. When it registers to the Asterisk server it is using the IP address on the private network and not the public IP of the NAT address. Can someone tell me what I need to do so the phone registerers using an internet address rather than the remote network NAT address. ___ --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] Polycom and NAT
In your sip.conf, for the user: nat=yes To send keepalives for the UDP connection (depending on how flimsy the device handling NAT is): qualify=yes From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C Sent: Tuesday, August 21, 2007 17:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom and NAT Hi All, I have a Polycom 501 that is behind a NAT. When it registers to the Asterisk server it is using the IP address on the private network and not the public IP of the NAT address. Can someone tell me what I need to do so the phone registerers using an internet address rather than the remote network NAT address. ___ --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] low-level dump for PRI dchan debugging
wanpipemon is the way to do it as far as I know. For starters, what do your zaptel/zapata configs look like? I would first verify that your D-channel is set properly, you can view that in the console as follows: asterisk pri show span 1/0 Primary D-channel: 24 Status: Provisioned, Up, Active Switchtype: National ISDN -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Anderson Sent: Monday, August 06, 2007 11:09 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] low-level dump for PRI dchan debugging I've been going back and forth with my telco for several days, trying different configurations to get a new PRI to come up. The bchannels are all up and the T1 is not in alarm status. The dchannel refuses to come up however. We've tried ni2, qsig, and now dms100 for the switchtype. The telco tech I've been working with says that he's been sending reset all channels signals to my system, to which he's getting an establish remote response from my asterisk box. I've been running a packet dump (wanpipemon -i w1g1 -c trd) of my d-channel this whole time and have yet to see a single incoming packet. I believe I *should* be seeing an incoming packet when he sends the reset, correct? Is there any way to do a completely raw dump of the d-channel? Here are my specs: linux-2.6.16 libpri-1.3.5 zaptel-1.2.19 asterisk-1.2.21.1 The PRI interface is a Sangoma A102...it's running the latest firmware and I'm running wanpipe-2.3.4-12 for the sangoma drivers. Any ideas? -- Erik Anderson http://andersonfam.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] low-level dump for PRI dchan debugging
Have you completely ignored the telco suggestion and attempted pri_cpe? Sounds like a miscommunication in settings to me. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Anderson Sent: Monday, August 06, 2007 12:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] low-level dump for PRI dchan debugging lpdlnx04*CLI pri show span 1 Primary D-channel: 24 Status: Provisioned, Down, Active Switchtype: Nortel DMS100 Type: Network I know it's odd, but the telco instructed me to set my equipment as the network end...hence pri_net: /etc/zaptel.conf loadzone=us defaultzone=us #Sangoma A102 port 1 [slot:10 bus:2 span: 1] span=1,1,0,esf,b8zs bchan=1-8 dchan=24 /etc/asterisk/zapata.conf [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;;Sangoma A102 port 1 [slot:10 bus:2 span: 1] switchtype=dms100 context=from-pstn group=1 signalling=pri_net channel = 1-8 There you go. As an aside, turns out that it's a national holiday in CA, so the Sangoma support guys are on vacation for the day. -erik ___ --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: Asterisk and COS bits
You have it right, for 1.2, use 'tos=', for 1.4 use 'tos_sip/tos_audio/tos_video'. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al lists Sent: Monday, July 23, 2007 10:21 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Fwd: Asterisk and COS bits Anyone? -- Forwarded message -- From: Al lists [EMAIL PROTECTED] Date: Jul 21, 2007 6:24 PM Subject: Asterisk and COS bits To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Is there any way to change COS bits for packets? There is a tos option on sip.conf, does asterisk change COS bits considering tos value? ___ --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] G729 copy protection
Make sure there are no other files in the license path other than your valid license for this server. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce McAlister Sent: Thursday, July 19, 2007 09:13 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] G729 copy protection Jared Smith wrote: On Thu, 2007-07-19 at 16:20 +0100, Bruce McAlister wrote: Am I doing something wrong? The README files dont quite explain how to get the Key-ID? You should have received a key from Digium when you bought your license to use the G.729 codec. If you haven't yet bought any G.729 licenses, you can buy them from Digium's website at http://www.digium.com/en/products/voice/g729codec.php OK, I got hold of the G729 Key that was issued to us by digium recently and have now successfully registered the codec on the host. However, it still comes back with the following warning on the console after a restart: [codec_g729a.so] = (Annex A/B (floating point) G.729 Codec (optimized for i686)) Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:465 load_module: G.729 transcoding module version 32, Copyright (C) 1999-2007 Digium, Inc. Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:469 load_module: This module is supplied under a commercial license granted by Digium, Inc. Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:470 load_module: Please see the full license text supplied by the accompanying Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:471 load_module: register utility, or ask for a copy from Digium. Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:472 load_module: This product includes software developed by the OpenSSL Project Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:473 load_module: for use in the OpenSSL Toolkit. (http://www.openssl.org/) Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:474 load_module: Copyright (C) 1998-2006 The OpenSSL Project Jul 19 17:07:27 WARNING[20591]: codec_g729.c:481 load_module: Failed to initialize G.729 copy protection! I can see the licence there (10 channel), but it looks like the codec does not want to inititalize properly. Thanks Bruce ___ --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] G729 copy protection
Correct, if you have multiple licenses in there (say a single storage location for a cluster of servers), it won't load. If you've tried other architectures of the codec and still had no luck, I'd say contact Digium support on it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce McAlister Sent: Thursday, July 19, 2007 13:55 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] G729 copy protection Darryl Dunkin wrote: Make sure there are no other files in the license path other than your valid license for this server. Hi, I have just checked this, and there is only the 1 license file in the /var/lib/asterisk/licenses directory. Is that what you meant? Thanks Bruce ___ --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] MultiParking
Look at app_valetparking here: http://www.voip-info.org/wiki/index.php?page=Asterisk+addons From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Kiely Sent: Monday, July 16, 2007 16:47 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] MultiParking Does anyone have the multiparking feature enabled in asterisk 1.4? or suggest multiple parking lots? ___ --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] G729 , upgrade asterisk
Licenses are stored in /var/lib/asterisk/licenses, not in the module itself. Won't need any reactivation between versions either. There is no real need to delete the modules folder between minor versions like this, 'make install' will overwrite the modules and warn you if there are any extra ones in there (it should always warn about the g729 module). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of AL Daei Sent: Saturday, June 30, 2007 18:12 To: asterisk-users@lists.digium.com Subject: [asterisk-users] G729 , upgrade asterisk I'm planning to upgrade my asterisk 1.4.4 to 1.4.6. usually for asterisk upgrade i delete modules directory and include, then compile the new version. Since i have couple of G729 Licenses on this server installed, would i need to call Digium to reactivate these Licenses? Is there any better and faster way of upgrade asterisk? Possibly without losing G729 License? Thanks! Play free games, earn tickets, get cool prizes! Join Live Search Club. Join Live Search Club! http://club.live.com/home.aspx?icid=CLUB_wlmailtextlink ___ --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] Bad Echo between SIP calls
This should only be for TDM to TDM calls, SIP to SIP calls don't use the zaptel driver. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Tuesday, June 12, 2007 16:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bad Echo between SIP calls I don't see this listed anywhere here in the replies so. In your zapata.conf file try changing: echocancelwhenbridged=no to: echocancelwhenbridged=yes ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bad Echo between SIP calls
What are the end devices? That seems to have been lost here. The real issue is the handsets as those are the devices introducing the echo (the only analog players here). Most likely a volume or gain issue on those handsets, what SIP devices are the echo issues between? If both people hear echo, both devices are at fault, if one person hears it, it is the other end at fault. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Deepak Naidu Sent: Tuesday, June 12, 2007 19:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Bad Echo between SIP calls I like the way people replied to this message of mine. It seems this thread is going back to the hybrid echo issue(no this is not the problem). As said by many ZAP is not in picture for SIP--SIP ie Ext-Ext internal calls. To put my inputs I did tons of QA on this issue to ground on whats the source. Its not just the phone or only the network but may be both. I am not sure how Asterisk would contribute to this. At time for a given 2 internal extension there was no echo but suddenly turned up. People dialing on my phone have echo but not on other at the same time I have few phones which I dial no echo. So ya dont know whats wrong. Thanks all for your inputs sharing ur experience. -- Deepak Darryl Dunkin [EMAIL PROTECTED] wrote: This should only be for TDM to TDM calls, SIP to SIP calls don't use the zaptel driver. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Tuesday, June 12, 2007 16:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bad Echo between SIP calls I don't see this listed anywhere here in the replies so. In your zapata.conf file try changing: echocancelwhenbridged=no to: echocancelwhenbridged=yes ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users What kind of emailer are you? Find out today - get a free analysis of your email personality. Take the quiz at the Yahoo! Mail Championship http://uk.rd.yahoo.com/mail/uk/taglines/default/championships/quiz/*htt p://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk/ . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bad Echo between SIP calls
The echo cancellation card is for SIP-Zap calls only, no echo cancellation is done in Asterisk for SIP only calls. SIP to SIP, media is just passed through the server untouched (using media flow through, which is the option in sip.conf of canreinvite=no) if you are not handling any translation, even when handling translation between SIP calls there shouldn't be any echo cancellation done in Asterisk for SIP only calls. The place to look at would be the remote SIP devices which is typically what is adding the echo, this is usually a gain issue of some sort depending on which handsets you are using. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Deepak Naidu Sent: Monday, June 11, 2007 16:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bad Echo between SIP calls Sounds crazy right? even was I, more over support guy logged in unloaded the zap modules to test them, still an echo. Ya, I was clear saying that we have SIP--- SIP issue ie internal extension echo problem. It seems the echo with SIP--SIP has many factors. I am just curios to eliminate any possibility of Asterisk failing to cancel the echo. OK, one question here howz the call flow when a SIP---SIP call is established ie. is the connection between 2 phones when an Internal call is made or does the SIP call goes via Asterisk once the SIP--SIP call is establised. -- Deepak Matthew Fredrickson [EMAIL PROTECTED] wrote: On Jun 8, 2007, at 6:00 PM, Deepak Naidu wrote: Hi, We have a PRI connection when its was on test networks we had echo problems withoutside line. So I bought a TE212P card resolve the echo problem. Which did to an extent. Its using asterisk 1.2.18 RHEL4-Update 4. But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. I am not sure whats the problem. Also there's slight echo when calling Digium support. Totally lost Digium says we need to remove the echo module to resolve SIP echo problems. Then ? the heck we pay for.. Are you sure that they understood that you were having this problem between 2 SIP endpoints? That advice only makes sense to test if one side is Zap and the other side is SIP. --- Matthew Fredrickson Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Answers - Get better answers from someone who knows. Try it now http://uk.answers.yahoo.com/;_ylc=X3oDMTEydmViNG02BF9TAzIxMTQ3MTcxOTAEc 2VjA21haWwEc2xrA3RhZ2xpbmU . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bad Echo between SIP calls
Best way to do this is not touch the sip.cfg, ever. Leave it as included in each release and include your overrides in a different file. Then reference your files like this in your MAC.cfg file, your file will override the sip.cfg defaults. CONFIG_FILES=phone_user.cfg,server.cfg,sip.cfg In server.cfg, if you wanted to change the server, for example: ?xml version=1.0 standalone=yes? sip voIpProt local voIpProt.local.port=/ server voIpProt.server.1.address=asterisk.yourdomain.com /voIpProt /sip From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Saturday, June 09, 2007 22:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bad Echo between SIP calls It doesn't matter if it's supported, they are all, however I have seen some echo problems after firmware upgrades, the only way to fix it was to either copy the differences or overwrite my old config files with the new ones that came with the firmware and then modify as needed for my setup. On 6/10/07, Deepak Naidu [EMAIL PROTECTED] wrote: The sip config firmware are the supported one for the existing firmware. If you have any stable working Polycom 501 SIP without echo between SIP--SIP wouldnt mind to share the sip.cfg, sip.ld bootrom would be great, bcos I have not got concreate resolution for this issue. Hope I can resolve this mess. Feels bad when one does best in aggregating things some louzy device screws up... Oh my frustation is comming on mail : http://us.i1.yimg.com/us.yimg.com/i/mesg/tsmileys2/03.gif -- Deepak C F [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Are the config files you are using with the phones what was meant with that firmware? or did you upgrade the firmware and reused the old config files? On 6/9/07, Steve Underwood wrote: Stephen Davies wrote: On 09/06/07, Deepak Naidu wrote: Ya, I have done that, below is zapata.conf . Also we had an TMP card with analog lines. SIP cals were great on them. now when we switched over. SIP calls have echo.. which shouldnt be at all. If you are getting echo on pure SIP to SIP calls, there's no point in fiddling around with your zapta.conf. That file is for configuring chan_zap, which is used to talk to Zap/ channels. Your calls are SIP to SIP so the zap channel and your PRI aren't being used at all. SIP calls are pure digital 4 wire lines so no electrical (Hybrid) echo will be present. The phones should not generate echo. If they are, they are presumably nasty phones (what kind are they?) and you should get properly made phones. By this measure most phones are nasty. The handset should be echo cancelled, to prevent leakage of the earpiece into the mike. It is getting less and less common to do this, now. Polycoms, Sipuras, Snoms, you name it, they do it badly. Many are not too annoying until someone turns the volume up. Call someone a little hard of hearing and you will hear echo. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at the Yahoo! Mail Championships http://uk.rd.yahoo.com/mail/uk/taglines/default/championships/games/*ht tp://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk/ . Plus: play games and win prizes. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Queue problems
Not only that, are the phones logged into the agents? The agents are most likely statically assigned but need to be logged into. This can be confusing. I use AddQueueMember/RemoveQueueMember for the phones themselves skipping the agents. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bkruse Sent: Friday, April 20, 2007 15:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue problems Are your agents logged into the queue? -brandon Tim Verscheure wrote: Hi, I've been configuring AsteriskNOW from the GUI but could it be that the GUI isn't working properly? because when I make a queue and add a few agents, and when I call the queue none of the phones ring. The queue is also configured at Ringall I checked the queues.conf file and the settings matched. I also noticed that the agents I made in the GUI, that they were not written away in agents.conf file, so I've added them there but still no results... any suggestions? Tim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] ztdummy does not load properly at server startup
It's not playing a wav file at all, it is mixing the live audio from all of the callers in that conference room and sending it back out to them. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Theo Band Sent: Thursday, April 19, 2007 13:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ztdummy does not load properly at server startup Or add /dev/null Why would one application need a special driver? What so different about the Meetme() application? Playing a wav file doesn't need a special timing source for instance. But, I'm just a simple end user of course, not understanding all the complex details of a PBX :-) Theo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] ${CALLERIDNUM} DEPRECATED in 1.4.2
${CALLERID(num)}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sanjay
Rajdev
Sent: Monday, April 16, 2007 13:39
To: asterisk-users
Cc: asterisk-dev
Subject: [asterisk-users] ${CALLERIDNUM} DEPRECATED in 1.4.2
${CALLERIDNUM} is DEPRECATED in 1.4.2 what is the alternative to find
the same in extensions.conf for setting a proper dialplan.
Please Suggest
Regards,
Sanjay Rajdev
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RE: [asterisk-users] queue report problem
You will probably find what you are looking for here: /var/log/asterisk/queue_log -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rilawich Ango Sent: Saturday, April 14, 2007 21:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] queue report problem HI all, I have a queue say 5000 and there are 10 member in the queue. When there is a call to the queue, the members will ring according to the defined strategy. In day end, I have to create a report about the queue and its member. But I found that it is very difficult to find the relation for the call to queue and the member who pick the call in CDR. Say, caller A calls the queue, queue member 9 pick the call. I want to know the caller A waiting time, conversion time for Caller A and member 9. Such relationship is very difficult to find in CDR. Anyone have such experience and how can I get such information? ango ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fax Blast over IP?
Either analog modems or a PRI, and Hylafax for automation, no VOIP involved there. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Thursday, April 12, 2007 10:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Fax Blast over IP? Basically, I want to send bulk faxes to a list of my clients. It is time consuming for a person to individually fax so a blast type solution seems best. Over IP is of course to save money... Thanks for the link, reading now... Any suggestions for the blast then? Wiley E. Siler Director of Information Technology Education 2020 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:[EMAIL PROTECTED] www.education2020.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Thursday, April 12, 2007 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax Blast over IP? On Thu, 12 Apr 2007, Wiley Siler said something to this effect: Can anyone recommend software that will allow me to utilize my VoIP provider and send fax over IP? Asterisk can send faxes, if you make it interoperate with a few well-known open-source utilities and/or software packages, depending on what precisely you want to do: http://www.voip-info.org/wiki-Asterisk+fax -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] ZAP device reference in Zaptel 1.4 - SIMILAR
After recompiling zaptel, did you recompile Asterisk? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bram kortleven Sent: Wednesday, April 04, 2007 14:51 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ZAP device reference in Zaptel 1.4 - SIMILAR Well, I'm experiencing a similar problem with my setup... debian etch, asterisk 1.4.2, zaptel 1.4.1, ... I cannot find the chan_zap.so module file anywhere, tried recompiling with zaptel 1.4.0... no change... I tried 'make menuselect', and going to the channels-part, chan_zap is marked XXX - dependencies missing: and this is the message for it, as an explanation. Zapata Telephony Depends on: zaptel_vldtmf(E), zaptel(E), tonezone(E) Can use: pri Anyone any idea how to resolve this?? Thanks On Wed, Apr 04, 2007 at 10:11:01AM +0300, Tzafrir Cohen wrote: Hi On Wed, Apr 04, 2007 at 04:59:41PM +1000, Devraj Mukherjee wrote: monk*CLI zap show channels No such command 'zap show' (type 'help' for help) Does that mean I dont have ZAP support in Asterisk? Maybe. ls -l /usr/lib/asterisk/modules/chan_zap.so I also repeat my second question: What is the contents of /etc/asterisk/zapata.conf ? Follow-up: The issue seems to be an issue with the atrpms package: http://bugzilla.atrpms.net/show_bug.cgi?id=1165 Asterisk 1.4.2 is missing chan_zap.so ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue12
November? It's DD/MM/ in his case, not MM/DD/. Either way, even two days is more than enough for me. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of john beaman Sent: Tuesday, April 03, 2007 12:43 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue12 I too was curious about this, so I copied the text into Babel Fish, and this is the result: I miss of the 2/04/2007 to the 11/04/2007. I will answer your message as of my return. For any urgency, to contact Emmanuelle Parache Moga or Cédric Buzay. If this guy is really going to be out until November these messages will get rather tiresome... John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom and Asterisk
I would be interested in specifics as I have yet to hear any real issues, a lot of people had some bad taste after 2.0.0, as is to be expected for a first release. I've used 2.0.2, 2.0.3, and now 2.1.0 with Asterisk 1.2 for months without issues. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Hammett Sent: Wednesday, March 28, 2007 14:30 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom and Asterisk I was previously having an issue with a Polycom phone and Polycom support said that Asterisk didn't play well with Polycom firmware versions 1.6.7 and newer due to SIP compatibility issues. I believe I heard a lot of things were fixed\adjusted in 1.4 and was wondering if anyone has had success with Asterisk 1.4 and the latest Polycom firmware releases. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom SoundPoint 501
What transport method are you using? Sounds like you are using DNSnaptr without specifying a port. When set to DNSnaptr, be sure you have both the hostname and port (5060) defined. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paolo Supino Sent: Wednesday, March 28, 2007 17:18 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom SoundPoint 501 Hi We've setup an Asterisk PBX recently and I encountered the following problem: When [mac address]-registration.cfg file includes the FQDN of the Asterisk PBX for the Polycom SoundPoint 501 phones it will not (even try to) register with the Asterisk PBX unless the DNS (it asks) successfully resolves the name: _sip._udp.[Asterisk FQDN]. Did this happen to anyone else? PS - The application version running on the SoundPoint 501s is 1.6.7.0098 TIA Paolo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Zaptel Dummy Driver
Question was off topic for the thread, but I'm feeling helpful today. More of a 1234... make install modprobe usb-uhci modprobe zaptel modprobe ztdummy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brad Sumrall Sent: Monday, March 19, 2007 13:17 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Zaptel silly issue I am geet this error, I assume because I have zero digium hardware installed. This is to be an entirely web based PBX. Can anyone point me to an easy 123 for installing zaptel in dummy form? I need music on hold for a VPS server. Brad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Zaptel Dummy Driver
Also forgot, ztdummy is not used with hold music, it would be used for mixing audio in the meetme app. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darryl Dunkin Sent: Monday, March 19, 2007 12:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Zaptel Dummy Driver Question was off topic for the thread, but I'm feeling helpful today. More of a 1234... make install modprobe usb-uhci modprobe zaptel modprobe ztdummy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brad Sumrall Sent: Monday, March 19, 2007 13:17 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Zaptel silly issue I am geet this error, I assume because I have zero digium hardware installed. This is to be an entirely web based PBX. Can anyone point me to an easy 123 for installing zaptel in dummy form? I need music on hold for a VPS server. Brad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] DST changes for the US
This all depends on the setting before it: tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 Since this isn't a fixed date, it isn't used the same way. It doesn't understand 'second week of the month', so if you use the 8th, it will use the next weekday of tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1. If start date was set to 2, it should change your clock on the 4th. Here are the working defaults from 2.1.0: tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.daylightSavings.start.date=8 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=11 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=0 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Monday, March 12, 2007 07:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DST changes for the US Peder @ NetworkOblivion wrote: I'm pretty sure this is wrong: tcpIpApp.sntp.daylightSavings.start.date=8 Should be: tcpIpApp.sntp.daylightSavings.start.date=2 This is what I set it to as well. 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP privacy headers
Look here: http://www.voip-info.org/wiki/view/P-Asserted-Identity+and+Remote-Party- ID+header From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Bishop Sent: Sunday, February 04, 2007 15:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP privacy headers Hi, Out ITSP has told us to user SIP privacy headers to hide outbound caller ID. Does anyone know how or if this can be done in Asterisk. I tried exten = s,3,SIPAddHeader(privacy=on) prior to executing Dial but no luck. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] How to Clone Asterisk
Assuming some defaults... your results may vary. /etc/asterisk = Configs /var/spool/asterisk = Voicemail, other spool files /var/lib/asterisk = Licenses (G729 for example), stock sounds, astdb, etc From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert DeVries Sent: Thursday, February 01, 2007 21:29 To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to Clone Asterisk I want to essentially transplant my existing Asterisk server to a new machine, and take the old sever out of service. Assuming I install Asterisk on the new machine, does anyone know what files I would have to copy over? What comes to mind are the *.conf files in /etc/asterisk, as well as the voicemail audio files. Anything else? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom Provistioning Issue
This is typically an error in one of your config files, either 0004f2023ecc.cfg or sip.cfg. What does your 0004f2023ecc.cfg file look like? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Walker Sent: Friday, January 26, 2007 12:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom Provistioning Issue From what I know this log show everything working Perfect. Then it goes to the Welcome screen then after a long time of processing, it errors out with a 0x1 error Any Ideas? 1005195711|so |4|00|-- Initial log entry -- 1005195711|so |4|00|+++ Note that bootrom log times are in GMT +++ 1005195711|hw |4|00|Initial log entry. 1005195711|wdog |4|00|Initial log entry 1005195711|cfg |4|00|Initial log entry 1005195711|copy |3|00|Initial log entry 1005195711|cdp |4|00|Initial log entry 1005195711|cdp |5|00|CDP is DISABLED. 1005195711|cdp |5|00|802.1Q/VLAN tagging is DISABLED. 1005195711|so |3|00|Platform: Model=SoundPoint IP 501, Assembly=2345-11500-040 Rev=A 1005195711|so |3|00|Platform: Board=2345-11500-040 A 1005195711|so |3|00|Platform: MAC=0004f2023ecc, IP=192.168.15.55, Subnet Mask=255.255.255.0 1005195711|so |3|00|Platform: BootBlock=2.5.0 (11500_040) 06-Nov-04 08:08 1005195711|so |3|00|Application, main: Label=BOOT, Version=3.2.2.0019 24-Aug-06 18:05 1005195711|so |3|00|Application, main: P/N=3150-11069-322 1005195711|app1 |4|00|Initial log entry. 1005195711|app1 |3|00|DNS resolver server is '192.168.15.10' 1005195711|app1 |3|00|DNS resolver search domain is '' 1005195711|app1 |3|00|Bootline: eim(0,0)bootHost:flash e=192.168.15.55:ff00 h=192.168.15.52 g=192.168.15.10 u=1 pw= tn=CircaIP 1005195712|so |3|00|Link status is Net up Speed 100 full Duplex, PC down. 1005195722|cfg |3|00|Beginning to provision phone 1005195722|copy |3|00|'ftp://1:[EMAIL PROTECTED]/bootrom.ld' from '192.168.15.52' 1005195722|cfg |3|00|Image bootrom.ld has not changed 1005195722|copy |3|00|Download of 'bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 1005195722|cfg |3|00|Downloaded bootROM is identical to Current version 3.2.2 1005195722|copy |3|00|'ftp://1:[EMAIL PROTECTED]/0004f2023ecc.cfg' from '192.168.15.52' 1005195722|copy |3|00|Download of '0004f2023ecc.cfg' succeeded on attempt 1 (addr 1 of 1) 1005195722|copy |3|00|'ftp://1:[EMAIL PROTECTED]/sip.ld' from '192.168.15.52' 1005195724|cfg |3|00|Image sip.ld has not changed 1005195724|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr 1 of 1) 1005195724|cfg |3|00|Downloaded application image is identical to current version 1005195724|cfg |3|00|Phone successfully provisioned 1005195752|app1 |4|00|Could not load time from 0.0.0.0(0.0.0.0). 1005195755|app1 |4|00|Loaded application sip.ld successfully, errors 0x0. 1005195755|app1 |6|00|Uploading boot log, time is THU OCT 05 19:57:55 2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom Provistioning Issue
Be sure that your mac.cfg file is pointing to a valid configuration file, I believe the 0x1 error is a missing file error. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Walker Sent: Friday, January 26, 2007 12:49 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom Provistioning Issue Fixed that issue but it does not change the error 0126204105|cfg |3|00|Image sip.ld has not changed 0126204105|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr 1 of 1) 0126204105|cfg |3|00|Downloaded application image is identical to current version 0126204105|cfg |3|00|Phone successfully provisioned 0126204136|app1 |4|00|Loaded application sip.ld successfully, errors 0x0. 0126204136|app1 |6|00|Uploading boot log, time is FRI JAN 26 20:41:36 2007 William M. Conlon wrote: Looks like the network time server isn't provisioned. -- Bill 1005195752|app1 |4|00|Could not load time from 0.0.0.0(0.0.0.0). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom Provistioning Issue
Looks alright there. The next config to check is where it loads your 'jason.cfg', any errors will be in your app logfile (as opposed to the boot one you pasted). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Walker Sent: Friday, January 26, 2007 13:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom Provistioning Issue ?xml version=1.0 standalone=yes? !-- Default Master SIP Configuration File-- !-- Edit and rename this file to Ethernet-address.cfg for each phone.-- !-- $Revision: 1.14 $ $Date: 2005/07/27 18:43:30 $ -- APPLICATION APP_FILE_PATH=sip.ld CONFIG_FILES=jason.cfg, sip.cfg MISC_FILES= LOG_FILE_DIRECTORY= OVERRIDES_DIRECTORY= CONTACTS_DIRECTORY=/ is my mac IP Darryl Dunkin wrote: This is typically an error in one of your config files, either 0004f2023ecc.cfg or sip.cfg. What does your 0004f2023ecc.cfg file look like? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Walker Sent: Friday, January 26, 2007 12:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom Provistioning Issue From what I know this log show everything working Perfect. Then it goes to the Welcome screen then after a long time of processing, it errors out with a 0x1 error Any Ideas? 1005195711|so |4|00|-- Initial log entry -- 1005195711|so |4|00|+++ Note that bootrom log times are in GMT +++ 1005195711|hw |4|00|Initial log entry. 1005195711|wdog |4|00|Initial log entry 1005195711|cfg |4|00|Initial log entry 1005195711|copy |3|00|Initial log entry 1005195711|cdp |4|00|Initial log entry 1005195711|cdp |5|00|CDP is DISABLED. 1005195711|cdp |5|00|802.1Q/VLAN tagging is DISABLED. 1005195711|so |3|00|Platform: Model=SoundPoint IP 501, Assembly=2345-11500-040 Rev=A 1005195711|so |3|00|Platform: Board=2345-11500-040 A 1005195711|so |3|00|Platform: MAC=0004f2023ecc, IP=192.168.15.55, Subnet Mask=255.255.255.0 1005195711|so |3|00|Platform: BootBlock=2.5.0 (11500_040) 06-Nov-04 08:08 1005195711|so |3|00|Application, main: Label=BOOT, Version=3.2.2.0019 24-Aug-06 18:05 1005195711|so |3|00|Application, main: P/N=3150-11069-322 1005195711|app1 |4|00|Initial log entry. 1005195711|app1 |3|00|DNS resolver server is '192.168.15.10' 1005195711|app1 |3|00|DNS resolver search domain is '' 1005195711|app1 |3|00|Bootline: eim(0,0)bootHost:flash e=192.168.15.55:ff00 h=192.168.15.52 g=192.168.15.10 u=1 pw= tn=CircaIP 1005195712|so |3|00|Link status is Net up Speed 100 full Duplex, PC down. 1005195722|cfg |3|00|Beginning to provision phone 1005195722|copy |3|00|'ftp://1:[EMAIL PROTECTED]/bootrom.ld' from '192.168.15.52' 1005195722|cfg |3|00|Image bootrom.ld has not changed 1005195722|copy |3|00|Download of 'bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 1005195722|cfg |3|00|Downloaded bootROM is identical to Current version 3.2.2 1005195722|copy |3|00|'ftp://1:[EMAIL PROTECTED]/0004f2023ecc.cfg' from '192.168.15.52' 1005195722|copy |3|00|Download of '0004f2023ecc.cfg' succeeded on attempt 1 (addr 1 of 1) 1005195722|copy |3|00|'ftp://1:[EMAIL PROTECTED]/sip.ld' from '192.168.15.52' 1005195724|cfg |3|00|Image sip.ld has not changed 1005195724|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr 1 of 1) 1005195724|cfg |3|00|Downloaded application image is identical to current version 1005195724|cfg |3|00|Phone successfully provisioned 1005195752|app1 |4|00|Could not load time from 0.0.0.0(0.0.0.0). 1005195755|app1 |4|00|Loaded application sip.ld successfully, errors 0x0. 1005195755|app1 |6|00|Uploading boot log, time is THU OCT 05 19:57:55 2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Multiple parking lot
There is an SVN branch with this feature: http://svn.digium.com/view/asterisk/team/oej/multiparking/ I had hope this would be a feature added to Asterisk 1.4, but fail to see it on the changelog. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron McCarthy Sent: Wednesday, January 24, 2007 21:00 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Multiple parking lot Hi list, Does anyone know any ways to have mutiple parking lots? I've got a pbx that 2 customers share, both need their own, and then have lights on the phone flash when they park the call (snom phones). Any ideals I'm not thinking of?!? Any help would be great! Thanks Ron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Some queries on g729 license.
per call means per terminating channel where encoding/decoding is required. Termination could be to translate to another codec (with another peer) or to Asterisk itself to handle menus, voicemail, conference calls. In the conference call setup, each caller uses a license. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Rubenstein Sent: Monday, January 08, 2007 12:56 To: Paul Cc: Asterisk-Users Subject: Re: [asterisk-users] Some queries on g729 license. All of which hassle and expense can be avoided by buying a license for Digium's codec, which is tested to work well with Asterisk (and might come with some support). And is pretty cheap per simul call. I wonder whether that per call means per codec instance, which could be multiple licenses on a single conference call, where multiple (even if not all) parties are getting de/encoded simultaneously. And whether there are other tools for editing (/mixing/transforming) g729 data, in realtime (streams) or not (files), and whether they require a license. Ideally sox or equivalent would work on g729, maybe with a codec plugin. On Mon, 2007-01-08 at 13:23 -0500, Paul wrote: First point to tackle in any case involving patent, copyright or trademark infringement is whether or not the infringing party would have been qualified to buy any usage rights at all. In a case where you license the Intel source(read the terms, it's not really that free), you would be applying for a license under some plan that includes certain minimum payments. Even if you wrote new source from scratch you would be in the same boat. Last time I looked at the plans, I didn't see anything with low minimums. So even if you wrote code from scratch and never used it on more than 6 channels, you might have done something that normally requires a large upfront payment. Use $10k as an example. In such a case owner of the patent might have an attorney initiate contact. If you are willing to communicate they might allow you to pay the minimum and be licensed. If you can't do that, they might offer a settlement where you stop using the codec and pay them some lesser amount. If the patent holder can easily prove the violation you might as well try to deal with them and get things settled fast. If you sell or give away the codec it is easier for them to dig up proof. If you have unhappy employees that might be the way they hear about the violation in the first place. Important consideration: Bankruptcy law generally excludes debts created by things like malicious or criminal acts. Matthew Rubenstein wrote: As far as I know, the g729 patent requires buying a license to operate any implementation of it, whether Digium's, Intel's, or any other. Digium is set up to collect royalties (perhaps at a favorable rate) as part of their license from the patent holder. I don't know about Intel or any other. Or what the mechanics are for enforcing the patent on someone who operates a codec without a license. On Mon, 2007-01-08 at 10:51 -0500, Al Bochter wrote: What about the free open source G729 Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: I connect to a PSTN carrier over SIP which requires me to connect with a g729 codec. I'm using them for just robocalling: Asterisk server originates calls which play a prerecorded file. Can I pre-encode those stored files in g729 so they don't need to be encoded for each call? If so, do I need a g729 license for each call, or just a license for the preencoder? If the robocalls accept incoming DTMF, do I need g729 licenses for those calls? On Mon, 2007-01-08 at 04:08 -0700, [EMAIL PROTECTED] wrote: Date: Mon, 08 Jan 2007 13:47:39 +0800 From: Leo Ann Boon [EMAIL PROTECTED] Subject: Re: [asterisk-users] Some queries on g729 license. To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Xue Liangliang wrote: Hi, all I am a pabx vendor from Singapore. Recently we are going to implement a failover solution for our customers using heartbeat, the asterisk server can failover perfectly, however the g729 codec canot work, because it is binded the mac address, we have bought two set of licenses, can you provide us some workaround for this scenario? It shouldn't be a problem if you're only doing IP takeover and have bound the licenses to each server separately. If you're sharing the storage, then that could pose a problem. Leo DatVoiz Singapore Pte Ltd -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com --
RE: [asterisk-users] Force re-read of sip.conf
Yes, a 'reload' will reload all configuration files. Instead, 'sip reload' should do what you want a little faster. Example: *CLI sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/sip_notify.conf': Found From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michelle Dupuis Sent: Thursday, November 30, 2006 16:30 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Force re-read of sip.conf I have an asterisk server with a dynamic public IP address. Once the IP changes, remote clients suddenly have one-way audio again. I can resolve the problem with a restart, but am thinking have adding a cron command which does this every night. Will a reload cause asterisk to respect the new IP address specified in sip.conf? Or do I have to restart? Thanks, MD ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Add Apps to Asterisk?
First, check for app_meetme.so in /usr/lib/asterisk/modules (wherever your modules path is). Next, in the CLI, do a 'show modules' to see if it is there. If not, check your modules.conf and add in 'load = app_meetme.so' assuming autoload is not enabled. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Rubenstein Sent: Tuesday, November 14, 2006 21:09 To: Asterisk-Users Subject: [asterisk-users] Add Apps to Asterisk? I've got an Asterisk (v1.2.11) installation running, but it doesn't seem to have the Meetme() app. At the CLI, I type Meetme , and it responds No such command 'Meetme'; meetme doesn't show up in CLI show modules . I'm running a SIP-only server at a datacenter where I can't add Digium (or any other) HW, and am running under CentOS. There is an /etc/asterisk/meetme.conf file, but I don't see anything to use it. What do I have to do, exactly, to install Meetme? What about the Conference command, or others not installed? I'd prefer to use the CentOS package system as much as possible, but I can compile source if necessary. Is there a HowTo on the Web somewhere that details this process? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Modprobe Zaptel
After running 'make install', do a 'depmod -a'. Then check /lib/modules for the file: find /lib/modules | grep zaptel Be sure the path/lib/modules/kernel/extra/zaptel.ko matches up with your currently running kernel (from uname-a) as that is where it will be checking. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian VaraniniSent: Thursday, November 09, 2006 15:21To: asterisk-users@lists.digium.comSubject: [asterisk-users] Modprobe Zaptel Hi,Can someone walk me through compiling and loading the Zaptel 1.2.10 driver for Mandriva 2006 kernel 2.6.12-12? When I compile and attempt a modprobe I get "module zaptel not found"ThanksJulian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Sticky Polycom 501 keys and handset
In Asterisk enter 'sip show peer name' and you can see this in the Useragent field. Example (for 2.0.1): Useragent : PolycomSoundPointIP-SPIP_501-UA/2.0.1.0313 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MikeSent: Tuesday, November 07, 2006 11:13To: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [asterisk-users] "Sticky" Polycom 501 keys and handset Disregard my previous message, I succeeded in downgrading my phones. And it worked, thanks Rick for the info. Is there any Polycom-specific mailing list I should be on to be aware of stuff like that? Also, would you know how to check the version of sip.ld remotely? I know how to reboot remotely, and I did for a few phones, but my paranoid self would like to double check and see if the sip.ld 1.6.7 re-installed ok by checking the current version. Is that even possible? Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rick SmithSent: November 7, 2006 11:28 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [asterisk-users] "Sticky" Polycom 501 keys and handset I had this EXACT same problem, and 2.0.x is the problem according to Polycom Tech Support. I had such a hard time explaining the problem, too Downgraded to 1.6.7 and all worked well again. Polycom says if youre using Asterisk, dont go past 1.6.7 until they say to. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MikeSent: Tuesday, November 07, 2006 11:02 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [asterisk-users] "Sticky" Polycom 501 keys and handset Hi, I've recently bought new Polycom 501 phones, upgraded to bootrom 3.2.2 and SIP 2.0.1. I just noticed something, which I first blamed on Asterisk and NATs (a 2 second silence at the beginning of a call). Something I'venoticed also on my old phone (which is having the same problem now, but its also been upgraded). My keys are sticky. Simple as that. Sometimes I press a number and the key comes up (the hardware seems fine) but the phone produces this lng tone as if I had pressed the key for 3 seconds. Even the receiver is sticky, giving my dialtone when I lift it only1-2 seconds after I lift the handset. It simply looks like the phone can't keep up, like a sluggishcomputer. Anybody has ever seem this? I'd like to downgrde to SIP 1.6.7 to see if the new sip app was the problem. How can I do that? I've placed the old sip.ld file where I had to, but the phone wont pick it up. Short of that, can somebody point me to the newest firmware (2.0.2) to see if thatwould help? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Sticky Polycom 501 keys and handset
This should be in your Asterisk sip_notify.conf file by default I believe (if not, add it with an appropriate name): [polycom-check-cfg]Event=check-syncContent-Length=0 Then in the Asterisk run this (assuming the phone is registered properly): sip notify polycom-check-cfg user If the configuration on your FTP server (assuming FTP/TFTP configuration) has changed, it will reboot. Otherwise, in your sip.cfg for your phones, look for this: voIpProt.SIP.specialEvent.checkSync.alwaysReboot="0" Change it to this to always reboot when receiving the notify: voIpProt.SIP.specialEvent.checkSync.alwaysReboot="1" From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rick SmithSent: Tuesday, November 07, 2006 17:44To: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [asterisk-users] "Sticky" Polycom 501 keys and handset hmm, Id like to know that. How do you reboot remotely ? J From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MikeSent: Tuesday, November 07, 2006 2:13 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [asterisk-users] "Sticky" Polycom 501 keys and handset Disregard my previous message, I succeeded in downgrading my phones. And it worked, thanks Rick for the info. Is there any Polycom-specific mailing list I should be on to be aware of stuff like that? Also, would you know how to check the version of sip.ld remotely? I know how to reboot remotely, and I did for a few phones, but my paranoid self would like to double check and see if the sip.ld 1.6.7 re-installed ok by checking the current version. Is that even possible? Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rick SmithSent: November 7, 2006 11:28 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [asterisk-users] "Sticky" Polycom 501 keys and handset I had this EXACT same problem, and 2.0.x is the problem according to Polycom Tech Support. I had such a hard time explaining the problem, too Downgraded to 1.6.7 and all worked well again. Polycom says if youre using Asterisk, dont go past 1.6.7 until they say to. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MikeSent: Tuesday, November 07, 2006 11:02 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [asterisk-users] "Sticky" Polycom 501 keys and handset Hi, I've recently bought new Polycom 501 phones, upgraded to bootrom 3.2.2 and SIP 2.0.1. I just noticed something, which I first blamed on Asterisk and NATs (a 2 second silence at the beginning of a call). Something I'venoticed also on my old phone (which is having the same problem now, but its also been upgraded). My keys are sticky. Simple as that. Sometimes I press a number and the key comes up (the hardware seems fine) but the phone produces this lng tone as if I had pressed the key for 3 seconds. Even the receiver is sticky, giving my dialtone when I lift it only1-2 seconds after I lift the handset. It simply looks like the phone can't keep up, like a sluggishcomputer. Anybody has ever seem this? I'd like to downgrde to SIP 1.6.7 to see if the new sip app was the problem. How can I do that? I've placed the old sip.ld file where I had to, but the phone wont pick it up. Short of that, can somebody point me to the newest firmware (2.0.2) to see if thatwould help? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
