[asterisk-users] Snom 320 with TDM02B and echo problems
Hello all, I instaled an asterisk 1.2.X, with two lines FXO and two snom 3200 phones, on an amd_64 processor. All goes well, the voice is clear on the remote side but in the Voip side, where the Snom 320 is placed, I hear my voice, but don't in the line, the echo is on the phone. I just play with zapata gain values and with the Snom mic volume, but the echos does not disapperars. the phone is updated to firmware 6.5.12, the last i have found. Any clue about how to eliminate de echo in the snom 320 phone? What could I do to solve that? Thanks in advance. VoipCrazy ___ --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] extensions.conf pattern match info
On Wed, Nov 07, 2007 at 11:03:30PM -0600, Eric ManxPower Wieling wrote: I'm not a fan of using the Wiki as a reference, but there really isn't any info like this in the docs that come with Asterisk. Agreed, except to note that the explanations given in Asterisk: The Future of Telephony serve as a good introduction to dial plans and, specifically, pattern matching. ___ --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] dtmfmode RFC2833 and inband
I have a grandstream 488 using FXO port. With asterisk 1.2.23 When I have DTMF mode set to RFC2833 (asterisk and grandstream) and I use a call file and AGI to originate the call I dont get the DTMF tones on the device. If I do it manually from a polycom 550 (set for RFC2833) it works. When I change the DTMF mode to inband on both asterisk and grandstream and use the call file it now works. If I now use the polycom 550 (still set for RFC2833) and manually call this does not work anymore - the grandstream is still set to inband now. What kind of issue am I having? Thanks, Jerry ___ --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] asterisk and installing chan_h323.so rpm
Hello, When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors. Failed dependencies: libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 But i found the same files in /usr/lib/libh323_linux_x86_r.so.1 /usr/lib/libpt_linux_x86_r.so.1 What to do for asterisk to detect the same files? Thanks Regards Bincy K 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
[asterisk-users] make h323 native transfer on stablished call
Hi all: I don't know if exist any other mailing more apropiated for this question. If exist, please let me know. I need orientation for this situation: 1. 1.4.13-BRIstuffed with support for h323 with asterisk-h323 module 2. An analog Pbx with support por h323 make asterisk a call, that asnwer and put with MOH 3. At this point I want asterisk to make a native h323 transfer of the current call to another h323 destination (out of asterisk), and after the transfer do not exist any h323 channel on the asterisk side. I know that is possible to make a 'natural' transfer inside asterisk. I mean, that asterisk make a new call to the h323 destination, and wen aswered make a bridge between two channels. But this isn't an option for me. I have to free resorces between the pbx and asterisk box. this is possible in h323? Is supported by the current channel implementation? 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] OT: Aastra 57i configuration via TFTP problem
Yes - we've been over this with Aastra support, and they acknowledge a bug in their firmware but can't seem to find it. They said wait for the next firmware release (and at least 2 releases have passed). We had SOME success by creating a blank config file, changing the order of entries in the config file, and reloading, resetting phone to factory defaults, etc. Enough playing and we got a config file good enough to go (but never could get all settings in). We do have some Aastra install where everything went great too. It's a nice phone, but for large deployments you run a real risk. We had one deployment where we had to swap out all Aastra for another phone because we had wasted 80 hours of staff time on firmware bugs, and frustrated with Aastra support. (We wasted more $ on staff time that it paid to just buy the client better phones). MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roi Stork Sent: Wednesday, November 07, 2007 10:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: Aastra 57i configuration via TFTP problem Here's what I did: 1) Reduced the local config to just network settings. 2) mac.cfg contains network + sip settings. 3) Restarted the phone. The result was only some sip settings like auth name, user name, password get updated. Fields such as proxy ip and registrar ip didn't get updated. I expected the whole sip settings were read from the cfg file and set, but it wasn't the case. Same problem happened to your setup? On Nov 7, 2007 6:33 PM, Michelle Dupuis [EMAIL PROTECTED] wrote: Use the web interface of the phone to retrieve the config file that you uploaded. Is it only partially there? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roi Stork Sent: Wednesday, November 07, 2007 9:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: Aastra 57i configuration via TFTP problem Thanks! We checked the TFTP server and there seems to be no problem. It's up and listening, and looking at the tcpdump and the log there really was traffic between the phone and the server. We also successfully downloaded files using another TFTP client. On Nov 7, 2007 5:33 AM, Jared Smith [EMAIL PROTECTED] wrote: On Wed, 2007-11-07 at 00:08 -0800, Roi Stork wrote: 1) No DHCP, so I manually set the network settings via phone UI. 2) The files aastra.cfg and mac address.cfg are in the TFTP root folder. 3) Restarted the phone. Here's what I'd do to troubleshoot the problem: 1) First make sure that your TFTP server is actually listening: [EMAIL PROTECTED] ~]# netstat --listen -npu | grep :69 udp0 0 0.0.0.0:690.0.0.0:* 2334/xinetd 2) Next, I'd use tcpdump to make sure you're actually seeing TFTP traffic from the phone: [EMAIL PROTECTED] ~]# tcpdump -vv port 69 tcpdump: listening on eth0, link-type EN10MB (Ethernet), capture size 96 bytes 08:28:56.622180 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 48) 192.168.0.100.34771 192.168.0.50.tftp: [udp sum ok] 20 RRQ test.txt netascii 1 packets captured 1 packets received by filter 0 packets dropped by kernel 3) As an additional step, you can turn up the verbosity of the tftp server, and look for it's messages in /var/log/messages. In my case, I simply add a -v to the server_args line in my /etc/xinetd.d/tftp file as show below and restart xinetd. server_args = -s /tftpboot -v The information shows up in /var/log/messages: Nov 7 08:28:56 hockey in.tftpd[27601]: RRQ from 192.168.0.100 filename test.txt Let me know if that helps, or if I need to go into more detail. -- Jared Smith Community Relations Manager Digium, Inc. ___ --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
[asterisk-users] Channel variables, any difference with SIP vs. IAX?
Hello, I have some extensions that are using variables loaded by an AGI program. Everything works fine and I am able to use NoOp to see the value of my variables when using IAX, but the same variables don't work when using SIP. I can provide further details, but right off of the bat does is there something I need to know about the use of user defined variables in with SIP channels vs. IAX channels? Thanks, Jason ___ --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] 'a' extension
Is there any way to see the called number when a call gets redirected to the 'a' extension from voicemail? Say x123 calls x456 and it rolls to voicemail. x123 hits * and gets dumped into the 'a' extension in the original context. I need some logic in 'a' to do a database lookup based on the original called number (x456). Any ideas? When I do a test, it appears that the called number is 'a' and the calling number is 123. I need to be able to tell that it was a call to x456. Thanks. Peder ___ --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] time on polycom 501
I have a polycom 501 phone that is 1 hour off now. Before last sunday (time change) the time was fine. ?xml version=1.0 standalone=yes? PHONE_CONFIG OVERRIDES _.0x20._log.level.change.sip=0 tcpIpApp.sntp.daylightSavings.stop.date=4 tcpIpApp.sntp.daylightSavings.stop.month=11 tcpIpApp.sntp.daylightSavings.start.date=8 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.gmtOffset=-18000 tcpIpApp.sntp.address=time.apple.com reg.1.ringType=4 lcl.cpt=0/ /PHONE_CONFIG I also have in dhcpd.conf: option ntp-servers 17.254.0.27; How can I get my polycom phones back to the correct time? Thanks, Jerry ___ --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] weird 185 secs timeout call problem
exten = whatever,1,Answer() rest of your dialplan for the queue Thanks, Steve Totaro Andre Quintaes wrote: On our tests using asterisk, some calls have been terminated abruptely with exact 185 seconds. This is happening with all our incoming calls from a trunk from 1 of my DID providers ( other providers or trunks are fine) and I could reproduce it by calling a queue from my Wengophone Softphone and letting the MoH play for 185 secs. If I make the same call from my WRTP54G on the same place, the call doest not get hung up after 185 secs. The incoming calls go trhough a queue and get mixmonitored. I will make further tests but I tried changing several timeout and keepalive parameters on sip.conf but nothing got effect. Even tried with reinvites enabled and disabled. Does any one have a clue? 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] time on polycom 501
Jerry Geis wrote: I have a polycom 501 phone that is 1 hour off now. Before last sunday (time change) the time was fine. Google is your friend: http://www.google.com/search?hl=enq=polycom+daylightSavingsbtnG=Google+Search Top hit fixed it for us. For the archives, in case the top hit is no longer the top hit (i.e. I'm feeling unlucky), the result is: http://knowledgebase.polycom.com/KanisaPlatform/Publishing/996/10627_f.SAL_PUBLIC_1_2.html and contains this: Description *Technical Bulletin 17803* *SoundPoint® and SoundStation® IP phones require configuration changes due to changes in daylight saving time (DST) dates.* *This information applies to:* • SoundPoint IP 300, 301, 430, 500, 501, 600, 601, 650 desktop phones and SoundStation IP 4000 conference phones *Note:* This information applies to the SoundPoint IP 650, where software releases exists to support the IP 650 (see Software Release Notes for platform compatibility). *SYMPTOMS* Beginning in 2006, all parts of the State of Indiana will observe Daylight Saving Time along with the rest of the United States. The majority of the state will now be in Eastern Time, but there are several counties near Chicago that will remain in Central Time. The United States Congress passed a law in 2005 that changes the dates when US Daylight Saving Time begins and ends starting in 2007. This affects all US states except Hawaii and Arizona, which do not observe DST. As of this writing, the Canadian provinces of Ontario, Manitoba, Quebec, Prince Edward Island, New Brunswick, Alberta, the Yukon and Northwest Territories, British Columbia, and Nova Scotia have indicated that they will adopt the same changes, and other provinces and territories will continue with current procedures. *RESOLUTION* With respect to the State of Indiana, no special configuration is required to support this change, but any special configuration that had been made previously to exempt phones in Indiana from DST needs to be removed. *Note:* The following change cannot be safely made until 6 November 2006, as the old settings are required until that date for 2006 DST to be calculated correctly. To configure phones for the new DST rules, the SNTP configuration section from sip.cfg (or ipmid.cfg in older versions) needs to change as follows: 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 *Note:* /These changes should be made in the *ipmid.cfg* configuration file for SoundPoint IP phones running the MGCP application./ *Note:* /There is an error in the display and setting of the DST ‘Start/Stop Day Of Week’ if the web server interface is used to set the DST rules. When the start date is set to 1 (Sunday) in the *sip.cfg* or *ipmid.cfg* file, it is displayed as Monday in the web server interface. If you use the web server interface to set the DST start/stop dates, select Monday to obtain a setting of Sunday. This discrepancy will be fixed in a future software release./ *STATUS* *Polycom recommends that this configuration change be made.* ___ --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] time on polycom 501
Hi Jerry, Here's what's in my SNTP tag: tcpIpApp.sntp.resyncPeriod=3600 tcpIpApp.sntp.address=192.168.15.50 tcpIpApp.sntp.gmtOffset=-18000 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 I had the same issue as you. The issue was the dayOfWeek.lastInMonth. For some reason I had set mine to 1. Digium has a KB article stating that it should be 0. Cheers, AR On Nov 8, 2007 10:46 AM, Jerry Geis [EMAIL PROTECTED] wrote: I have a polycom 501 phone that is 1 hour off now. Before last sunday (time change) the time was fine. ?xml version=1.0 standalone=yes? PHONE_CONFIG OVERRIDES _.0x20._log.level.change.sip=0 tcpIpApp.sntp.daylightSavings.stop.date=4 tcpIpApp.sntp.daylightSavings.stop.month=11 tcpIpApp.sntp.daylightSavings.start.date=8 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.gmtOffset=-18000 tcpIpApp.sntp.address=time.apple.com reg.1.ringType=4 lcl.cpt=0/ /PHONE_CONFIG I also have in dhcpd.conf: option ntp-servers 17.254.0.27; How can I get my polycom phones back to the correct time? Thanks, Jerry ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --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] 'a' extension
On 11/8/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: Is there any way to see the called number when a call gets redirected to the 'a' extension from voicemail? Say x123 calls x456 and it rolls to voicemail. x123 hits * and gets dumped into the 'a' extension in the original context. I need some logic in 'a' to do a database lookup based on the original called number (x456). Any ideas? When I do a test, it appears that the called number is 'a' and the calling number is 123. I need to be able to tell that it was a call to x456. Thanks. You're sending them into VoiceMail() from your dialplan - just stick the dialed number in a channel var before calling VoiceMail(), then refer to it in your 'a' extension. -- j. ___ --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] Channel variables, any difference with SIP vs. IAX?
Off-hand, have you compared the output of agi debug (on the console) between the working and non-working calls? I believe the variables all get displayed. -jr On Nov 8, 2007 6:13 AM, Jason Wolfe [EMAIL PROTECTED] wrote: I have some extensions that are using variables loaded by an AGI program. Everything works fine and I am able to use NoOp to see the value of my variables when using IAX, but the same variables don't work when using SIP. I can provide further details, but right off of the bat does is there something I need to know about the use of user defined variables in with SIP channels vs. IAX channels? -- Josh Richards - Grover Beach, California US [EMAIL PROTECTED] (don't forget the middle 't' initial when writing) http://blog.joshrichards.org/ 805/471-6923 (cell) Geek Research (Technology Management Consulting) - http://www.geekresearch.com/ Support These Nifty Causes: http://Kiva.org http://RoomToRead.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
Re: [asterisk-users] SIP: To: header?
7 nov 2007 kl. 14.26 skrev Tony Mountifield: Quick question for those who know the innards of chan_sip: Does chan_sip use the To: header of an incoming INVITE request, for anything other than setting SIP_HEADER(TO) ? No. Like e-mail software not using the To: header in the actual e-mail. As far as I can tell so far, the target extension is taken from the request URI, i.e. sip:[EMAIL PROTECTED], and the target context is taken from the section in sip.conf that matches the request's source IP address. Is that correct? Or by matching a user section by From: username. /O ___ --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] 'a' extension
I think you can save/get the number in variable and then assign it to callerid. I am doing similar and working for me. Thanks, Viv On 11/8/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: Is there any way to see the called number when a call gets redirected to the 'a' extension from voicemail? Say x123 calls x456 and it rolls to voicemail. x123 hits * and gets dumped into the 'a' extension in the original context. I need some logic in 'a' to do a database lookup based on the original called number (x456). Any ideas? When I do a test, it appears that the called number is 'a' and the calling number is 123. I need to be able to tell that it was a call to x456. Thanks. Peder ___ --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] 'a' extension
Peder @ NetworkOblivion wrote: Is there any way to see the called number when a call gets redirected to the 'a' extension from voicemail? Say x123 calls x456 and it rolls to voicemail. x123 hits * and gets dumped into the 'a' extension in the original context. I need some logic in 'a' to do a database lookup based on the original called number (x456). Any ideas? When I do a test, it appears that the called number is 'a' and the calling number is 123. I need to be able to tell that it was a call to x456. Thanks. You could set a CDR variable called origdst when the call starts up and then use the customer CDR format to kick origdst back out into the custom CDR format when the record gets written. This is what we do for this. -- -- Bird's The Word Technologies, Inc. http://www.btwtech.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
Re: [asterisk-users] __sip_xmit problem
Dialing Exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]) is valid. If you can post your dial plan and we can take a look (though I have never seen this error before). - Original Message - From: Rizwan Hisham [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, November 02, 2007 3:32 PM Subject: Re: [asterisk-users] __sip_xmit problem Hi, There seems to be only one suspecious thing in your dial command which is @10.0.0.22. Why are you using IP address in dial command. I dont know if its allowed in a dial command as an argument or not. I have checked, all the other arguments in your dial command are fine. Try dilaing without the IP address. Hope it solves your problem. Regards On Nov 2, 2007 9:05 AM, Rilawich Ango [EMAIL PROTECTED] wrote: Hi, I got the following warning from CLI when I try to execute the Dial command. It makes the call failed. Anyone can tell me what does it mean and how to solve? -- Executing [EMAIL PROTECTED]:61] Dial(SIP/4009-1f178ba0, SIP/[EMAIL PROTECTED]|35|L(7200:12)) in new stack -- Limit Data for this call: timelimit = 7200 play_warning = 12 play_to_caller = yes play_to_callee = no warning_freq = 0 start_sound= (null) warning_sound = timeleft end_sound = (null) [Nov 2 11:54:00] WARNING[8218]: chan_sip.c:1775 __sip_xmit: sip_xmit of 0x1f188330 (len 793) to 10.0.0.22:0 returned -1: Invalid argument -- Called [EMAIL PROTECTED] [Nov 2 11:54:01] WARNING[8206]: chan_sip.c:1775 __sip_xmit: sip_xmit of 0x1f188330 (len 793) to 10.0.0.22:0 returned -1: Invalid argument [Nov 2 11:54:02] WARNING[8206]: chan_sip.c:1775 __sip_xmit: sip_xmit of 0x1f188330 (len 793) to 10.0.0.22:0 returned -1: Invalid argument ___ --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 -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.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] Client lost on skinny
Paul wrote: I have six cisco 7911g connected on asterisk over chan_skinny. Four of them are working OK. two of them even the screen on the phone is indicating that is registered and has number loose connection to asterisk . On asterisk the message is Skinny Client was lost, unregistering. also this phones does not appear anymore in the skinny show devices list . If I dial the tone does not stop asterisk and i get a message like Asked to transmit on a non existent session . Can somebody help me ? What version of Asterisk? Registration tracking and recovery was reworked around version 1.4.7 or 1.4.8 If you have a version newer than that, what value are you using in skinny.conf for the keepAlive setting? Dan ___ --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 Queue Issue
Setup a 2nd registration on the phone that only allows 1 call at a time. Ideal setup it up as a shared appearance so call forwarding, etc dont work on that registration. This way your phone has 2 registrations 1 for any direct call and another for shared calls, queues, etc. ___ --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 1.4 + Presence
* hint: The 'hint' priority associates an extension with an Asterisk channel for the purpose of mapping the state of the channel to a state of the extension. In asterisk, a channel (technology/device) can have several states (unavailable, in-use, busy, ringing, etc) but an extension is just a label for a sequence of applications. However, when communicating the state of the channel to an external device, such as a receptionist console, you cannot use the Asterisk internal channel names, but must use an externally identifiable resource name, typically the extension number. A device would then subscribe to the state of the extension of interest and receive status notifications from the supporting technology channel. This is used in the SIP channel (implemented via the SUBSCRIBE/NOTIFY mechanism of RFC-3265) to light up the status lamps on SIP phones. This is supported in SNOM phones (see also) with their programmable keys set to type destination, as well as in Polycom (500/600), Aastra ( 480i, 9133i ), and Sayson phones. It is also supported in Citel SIP Handset Gateways. Privacy considerations: In sip.conf you can define a subscribecontext= value that determines in which context Asterisk should search for the matching extension when a subscribe request is received from the phone; however, if the extension doesn't exist in that context Asterisk is going to look for it in the default context! In other words: Everyone can subscribe to a hinted extension that is defined in the default context. By the way, specifying an empty subscribecontext is also fine if the phone should not at all subscribe to _any_ context. Likewise bug/patch 5515 (post Asterisk 1.2.0) adds devstate support also for MGCP (so far SIP, IAX and ZAP are supported; show channeltypes tell you which channels in your Asterisk support device status notification). Question: Does this patch only show a device which is unavailable (e.g. disconnected), or does it also show busy? Answer: Also busy (in use). Also chan_capi-cm v0.6.2 and later comes with basic hint support. It appears, however, that the dynamic naming of CAPI channels that includes the called number makes monitoring of a CAPI line for outgoing calls practically impossible - at least for now. Note: the 3rd party Bristuff patches come with app_devstate that permits state manipulation through the dialplan. New: While Asterisk 1.6 will include func_devstate natively there is now also a backport available for 1.4. This is quite similar to app_devstate as part of the bristuff patches. Example exten = 200,hint,SIP/phone1 ; this is case sensitive (!) in 1.0.9 and 1.2.0 exten = 200,1,Macro(stdexten,SIP/phone1) If you want to monitor the state of multiple phones using one speeddial, you can do so: exten = 200,hint,SIP/201SIP/202SIP/203 Asterisk seems to provide syntax for allowing more than one channel to be mapped to any particular extension with the hint system. Useful CLI commands for debugging are SIP show subscriptions, show hints, show channeltypes and SIP show inuse. On Nov 6, 2007 1:36 PM, Alejandro Cabrera Obed [EMAIL PROTECTED] wrote: Dear all, I'm using Asterisk 1.4 as my SIP server of my LAN users. The SIP clients are using different operating systems such Debian, Gentoo and Windows XP so they use different SIP softphones like SJPhone, Twinkle and X-Lite. In order to let SIP clients to see the presence status to each other, do I have to establish any special setting in Asterisk 1.4 ??? Or the presence status (online, offline, away, etc.) is only up to the SIP clients and not up to the Asterisk ??? Really thanks Alejandro ___ --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] dtmf / misdn
On Tue, 2007-11-06 at 22:16 -0800, Josh Richards wrote: This may be what you need: http://www.misdn.org/index.php/FAQ#Why_are_my_dtmf_tones_not_detected_everytime.3F Also, something here may be helpful: http://www.voip-info.org/wiki/view/Asterisk+DTMF#Troubleshooting -jr On Nov 6, 2007 2:12 PM, Hans Witvliet [EMAIL PROTECTED] wrote: Hi all, Perhaps someone can give me a hint i the right direction... Sometimes dtmf is recognized, sometimes not. I'm using 1.2.19 asterisk with misdn for my hfc card. When i got in incoming sip-call, dtmf is recognized, When i phone my self (isdn-phone or gsm-phone) no problem with dtmf When SOME (not all) people phone me (isdn-incoming) DTMF is not recognized. How come? Either it works for a particular configuration, or it doesn't. It doesn't make sense to me that it works sometimes... Thanx, just reduced dtmfthreshold to 50. Have to wait see if it does the trick. Funny thing is, that it effect just people using a shabby telco-provider ;) HW ___ --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 Help
On Nov 6, 2007 2:25 PM, Jarga Jallow [EMAIL PROTECTED] wrote: Under asterisk info: Sip registry 12/12 76.xxx.xxx.xxx D N 5066 UNREACHABLE 11/11 76.xxx.xxx.xxx D N 5064 UNREACHABLE 10/10 76.xxx.xxx.xxx D N 5062 UNREACHABLE All these IP phones are behind NAT. What could be the problem? You aren't supposed to be registering to your IP phones you should have the IP phones registering against your Asterisk. ___ --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] If caller id is null set to a specific number
All, If someone calls into my asterisk box and has a private number I would like to set the callers id to a specific telephone number, only when the ANI is missing, otherwise if present just pass it along. Any ideas? TIA, Jon___ --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 as a SIP to XMPP Jingle voice gateway
Openfire has a SIP soft phone plugin (and an AsteriskIM plugin - different functionality though) http://www.igniterealtime.org/ Eric Chamberlain wrote: Hello, Im looking for a SIP to XMPP Jingle voice gateway. I see that Asterisk has Jabber and Jingle support, but it looks like Asterisk acts as a Jabber client. Are there any Jabber server solutions, where Jabber users can call SIP users by using the SIP URI and vice versa? -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.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] If caller id is null set to a specific number
Have a look at the smartCID script on www.generationt.com It allows you to have a database of numbers and override the name (and number), flag numbers for screening, etc. MD _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Weisman Sent: Thursday, November 08, 2007 7:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] If caller id is null set to a specific number All, If someone calls into my asterisk box and has a private number I would like to set the callers id to a specific telephone number, only when the ANI is missing, otherwise if present just pass it along. Any ideas? TIA, Jon ___ --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] Route an incoming call by ANI*DNIS
Dan, What happen w/ this? Did you figure it out? I've setup w/ XO using ESF/B8ZS, and 5ESS for the switchtype, worked great and got the ANI as well. I dont think you can get ANI on EM Wink trunks, how about feature group d? -Jon - Original Message - From: Dan Casey [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, November 02, 2007 9:47 AM Subject: [asterisk-users] Route an incoming call by ANI*DNIS does anyone know how to route a call coming in with ANI*DNIS* Oct 31 12:47:41 VERBOSE[30056] logger.c: -- Executing Set(Zap/49-1, DID=1231234*4812*) in new stack I tried making a route for _.*4812* but that matched everything rather then just the dnis i wanted.. any ideas? I would preferably like pass the callerid along to my extensions, but for now the important thing is routing. 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] Asterisk and OBDC
Hi there! Hein Maxi! Don't hate me for this hâ?!?!? Well, I think that the subject is just right for asterisk... I need access a base on MS Access, to retrive some status, like just Yes or No. Just it!!! I think to work with ODBC in this case, and use the app_odbc... I try found some trip on Google, but all I get is about cdr on ODBC... I need more then that... What you think about? Thanks so much for any help... - Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! ___ --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] AsteriskNOW - how to open SIP ports?
Hi all, We're running AsteriskNOW Beta 6, and port 5060 is closed. We've checked it with a port scanner and the port is definitely closed. How do we open that port, either through the GUI or CLI. Is there any way to access the linux command line through AsteriskNOW? Thanks in advance! --Zaheer ___ --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] Client lost on skinny
Paul wrote: Thank you for your answer. I am using asterisk 1.4.13 and keepalive has a value of 120 in skinny.conf. You can try reducing the keepAlive. The phone will still loose registration, but will re-register faster. Other than that, I would look at the health of your network, especially the ports for the phones that are dropping off. Dan ___ --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] Cisco IP Communicator with Asterisk
I'm not sure if anyone has done this before or not but, I was able get the Cisco IP Communicator soft phone to work with Asterisk using SIP. Thought I would share my experiences. The key is on the installation. To have the software use the SIP protocol type the following command: msiexec /i CiscoIPCommunicatorSetup.msi /qb SIP=1. After installation configuration is just like configuring a Cisco 7970 hard phone. I used the configuration instructions outlined by Kerry Garrison at Asterisk Tutorials http://www.asterisktutorials.com/showproduct.php?ProductID=10. Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ --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] Wifi handover/roaming
For fastest handover disable any sort of encryption and use the same SSID for all AP... infact I don't know how you would setup roaming otherwise. Channels don't have to be the same, but optimize for the best RF performance/least channel overlap. ___ --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] If caller id is null set to a specific number
I get the same response with or w/o ANI... :(
- Original Message -
From: Doug Lytle [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, November 08, 2007 8:14 PM
Subject: Re: [asterisk-users] If caller id is null set to a specific number
Jon Weisman wrote:
All,
If someone calls into my asterisk box and has a private number I would
like to set the callers id to a specific telephone number, only when
the ANI is missing, otherwise if present just pass it along. Any ideas?
[incoming]
exten = s,1,Gosubif($[${CALLERID(number)} = ]?set-cid,s,1:2)
[set-cid]
exten = s,1,Set(CALLERID(number)=5551212)
exten = s,n,Return()
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
___
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___
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Re: [asterisk-users] Asterisk as a SIP to XMPP Jingle voice gateway
Openser has SIP-to-XMPP Gateway and JABBER IM and PRESENCE interconnection modules. I used the XMPP module with Google Talk and Asterisk a while back. It resulted in a segmentation fault, but again that was a long time ago. Eric Chamberlain wrote: Hello, I’m looking for a SIP to XMPP Jingle voice gateway. I see that Asterisk has Jabber and Jingle support, but it looks like Asterisk acts as a Jabber client. Are there any Jabber server solutions, where Jabber users can call SIP users by using the SIP URI and vice versa? -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.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] Asterisk and OBDC
2007/11/8, Gilberto Nunes [EMAIL PROTECTED]: Hi friends I have an application that storage data in MS Access. I need some library that can read the status of data in this base... Some one can help me?!?!?! Thanks Gilberto, what does your post has to do with Asterisk? Anyway, probably you can carry the info into a MySQL database and then work with it as you want Maxi ___ --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] AST-2007-024 - Fallacious security advisory spread on the Internet involving buffer overflow in Zaptel's sethdlc application
Asterisk Project Security Advisory - AST-2007-024 ++ | Product | Zaptel| |+---| | Summary | Potential buffer overflow from command line | || application sethdlc | |+---| | Nature of Advisory | Buffer overflow | |+---| | Susceptibility | Local sessions| |+---| | Severity | None | |+---| | Exploits Known | None | |+---| |Reported On | October 31, 2007 | |+---| |Reported By | Michael Bucko michael DOT bucko AT eleytt DOT| || com | |+---| | Posted On | October 31, 2007 | |+---| | Last Updated On | November 1, 2007 | |+---| | Advisory Contact | Mark Michelson mmichelson AT digium DOT com | |+---| | CVE Name | CVE-2007-5690 | ++ ++ | Description | This advisory is a response to a false security | | | vulnerability published in several places on the | | | Internet. Had Asterisk's developers been notified prior | | | to its publication, there would be no need for this. | | | | | | There is a potential for a buffer overflow in the| | | sethdlc application; however, running this application | | | requires root access to the server, which means that | | | exploiting this vulnerability gains the attacker no more | | | advantage than what he already has. As such, this is a | | | bug, not a security vulnerability. | ++ ++ | Resolution | The copy of the user-provided argument to the buffer has | || been limited to the length of the buffer. This fix has| || been committed to the Zaptel 1.2 and 1.4 repositories,| || but due to the lack of severity, new releases will not be | || immediately made. | || | || While we appreciate this programming error being brought | || to our attention, we would encourage security researchers | || to contact us prior to releasing any reports of their | || own, both so that we can fix any vulnerability found | || prior to the release of an announcement, as well as | || avoiding these types of mistakes (and the potential | || embarrassment of reporting a vulnerability that wasn't) | || in the future.| ++ ++ | Affected Versions| || | Product | Release Series | | |-++-| | Zaptel | 1.2.x | All versions prior to 1.2.22|
Re: [asterisk-users] If caller id is null set to a specific number
Jon Weisman wrote:
All,
If someone calls into my asterisk box and has a private number I would
like to set the callers id to a specific telephone number, only when
the ANI is missing, otherwise if present just pass it along. Any ideas?
[incoming]
exten = s,1,Gosubif($[${CALLERID(number)} = ]?set-cid,s,1:2)
[set-cid]
exten = s,1,Set(CALLERID(number)=5551212)
exten = s,n,Return()
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
___
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[asterisk-users] Asterisk and OBDC
Hi friends I have an application that storage data in MS Access. I need some library that can read the status of data in this base... Some one can help me?!?!?! Thanks -- Gilberto Nunes Itajaí - SC ___ --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] Client lost on skinny
Hi Dan, Thank you for your answer. I am using asterisk 1.4.13 and keepalive has a value of 120 in skinny.conf. 2007/11/8, Dan Austin [EMAIL PROTECTED]: Paul wrote: I have six cisco 7911g connected on asterisk over chan_skinny. Four of them are working OK. two of them even the screen on the phone is indicating that is registered and has number loose connection to asterisk . On asterisk the message is Skinny Client was lost, unregistering. also this phones does not appear anymore in the skinny show devices list . If I dial the tone does not stop asterisk and i get a message like Asked to transmit on a non existent session . Can somebody help me ? What version of Asterisk? Registration tracking and recovery was reworked around version 1.4.7 or 1.4.8 If you have a version newer than that, what value are you using in skinny.conf for the keepAlive setting? Dan ___ --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: To: header?
Ah, but is EXTREMELY useful for people like me who need to know where the call was originally destined for when it bounces off a Centrex DMS-100 due to call forward no answer or call busy condition. I can treat it as the DID for the original number and then send the caller into VM or other location as if it were the DID actually assigned to me. :) Johansson Olle E wrote: 7 nov 2007 kl. 14.26 skrev Tony Mountifield: Quick question for those who know the innards of chan_sip: Does chan_sip use the To: header of an incoming INVITE request, for anything other than setting SIP_HEADER(TO) ? No. Like e-mail software not using the To: header in the actual e-mail. As far as I can tell so far, the target extension is taken from the request URI, i.e. sip:[EMAIL PROTECTED], and the target context is taken from the section in sip.conf that matches the request's source IP address. Is that correct? Or by matching a user section by From: username. /O ___ --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] Snom 320 with TDM02B and echo problems
On Thu, Nov 08, 2007 at 10:22:41AM +0100, voip crazy wrote: I instaled an asterisk 1.2.X, with two lines FXO and two snom 3200 phones, on an amd_64 processor. All goes well, the voice is clear on the remote side but in the Voip side, where the Snom 320 is placed, I hear my voice, but don't in the line, the echo is on the phone. I just play with zapata gain values and with the Snom mic volume, but the echos does not disapperars. the phone is updated to firmware 6.5.12, the last i have found. Mine came with 7.1.8. Perhaps you should contact Snom to find out whether you can obtain the new version of the firmware. I suspect the problem is more likely to be on the POTS side of the connection however. ___ --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] Switchvox Space Requirements
Can anybody give me a rough idea how much disk space is requiered for a typical install? I want to install it in a system with solid state storage and I don't want to buy more than I need. Would 1GB be enough? Thanks, -- Andres ___ --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] Asterisk as a SIP to XMPP Jingle voice gateway
Hello, I'm looking for a SIP to XMPP Jingle voice gateway. I see that Asterisk has Jabber and Jingle support, but it looks like Asterisk acts as a Jabber client. Are there any Jabber server solutions, where Jabber users can call SIP users by using the SIP URI and vice versa? -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.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
[asterisk-users] Wanted: tutorial on troubleshooting SIP issues
For someone that's network-aware, but hasn't sat down and plowed through umpteen SIP-related RFC's and memorized the standards, is there a good primer on troubleshooting SIP issues? I'm seeing a lot of NOTIFY/603 messages on my network between Asterisk and my Sipura 942's, for instance... Not sure what these are... perhaps the qualify keepalives? In which case, I guess the 603 is moot... but since the messages are originating from the Sipuras to Asterisk and not vice-versa, it wouldn't seem to be the qualify... Next guess would be that they're NAT keepalives, but Asterisk and the phones are on the same private subnet (which in turn *is* NATted)... Anyway, pointers for someone wanting to learn to quickly diagnose SIP conversations would be great. Thanks, -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
Re: [asterisk-users] OT: Aastra 57i configuration via TFTP problem
Problem solved. You need to remove the config (local.cfg) that resides on the phone, before restarting it. I used the Web UI for awhile to tinker with the XML softkeys and SIP config. What happens is that the input was set into the local.cfg. The admin manual says that there is a precedence rule on how the phone reads the config: aastra.cfg, mac.cfg and finally local.cfg. The next cfg overwrites the values of the previous. That's why when I switched to TFTP, the values in mac.cfg get overwritten by local.cfg. Removing/Clearing the fields in the Web UI + restarting the phone won't work either, because the parameters (sip proxy ip, softkey etc) will still be there, this time with blank values. On Nov 8, 2007 6:58 AM, Michelle Dupuis [EMAIL PROTECTED] wrote: Yes - we've been over this with Aastra support, and they acknowledge a bug in their firmware but can't seem to find it. They said wait for the next firmware release (and at least 2 releases have passed). We had SOME success by creating a blank config file, changing the order of entries in the config file, and reloading, resetting phone to factory defaults, etc. Enough playing and we got a config file good enough to go (but never could get all settings in). We do have some Aastra install where everything went great too. It's a nice phone, but for large deployments you run a real risk. We had one deployment where we had to swap out all Aastra for another phone because we had wasted 80 hours of staff time on firmware bugs, and frustrated with Aastra support. (We wasted more $ on staff time that it paid to just buy the client better phones). MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roi Stork Sent: Wednesday, November 07, 2007 10:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: Aastra 57i configuration via TFTP problem Here's what I did: 1) Reduced the local config to just network settings. 2) mac.cfg contains network + sip settings. 3) Restarted the phone. The result was only some sip settings like auth name, user name, password get updated. Fields such as proxy ip and registrar ip didn't get updated. I expected the whole sip settings were read from the cfg file and set, but it wasn't the case. Same problem happened to your setup? On Nov 7, 2007 6:33 PM, Michelle Dupuis [EMAIL PROTECTED] wrote: Use the web interface of the phone to retrieve the config file that you uploaded. Is it only partially there? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roi Stork Sent: Wednesday, November 07, 2007 9:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: Aastra 57i configuration via TFTP problem Thanks! We checked the TFTP server and there seems to be no problem. It's up and listening, and looking at the tcpdump and the log there really was traffic between the phone and the server. We also successfully downloaded files using another TFTP client. On Nov 7, 2007 5:33 AM, Jared Smith [EMAIL PROTECTED] wrote: On Wed, 2007-11-07 at 00:08 -0800, Roi Stork wrote: 1) No DHCP, so I manually set the network settings via phone UI. 2) The files aastra.cfg and mac address.cfg are in the TFTP root folder. 3) Restarted the phone. Here's what I'd do to troubleshoot the problem: 1) First make sure that your TFTP server is actually listening: [EMAIL PROTECTED] ~]# netstat --listen -npu | grep :69 udp0 0 0.0.0.0:690.0.0.0:* 2334/xinetd 2) Next, I'd use tcpdump to make sure you're actually seeing TFTP traffic from the phone: [EMAIL PROTECTED] ~]# tcpdump -vv port 69 tcpdump: listening on eth0, link-type EN10MB (Ethernet), capture size 96 bytes 08:28:56.622180 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 48) 192.168.0.100.34771 192.168.0.50.tftp: [udp sum ok] 20 RRQ test.txt netascii 1 packets captured 1 packets received by filter 0 packets dropped by kernel 3) As an additional step, you can turn up the verbosity of the tftp server, and look for it's messages in /var/log/messages. In my case, I simply add a -v to the server_args line in my /etc/xinetd.d/tftp file as show below and restart xinetd. server_args = -s /tftpboot -v The information shows up in /var/log/messages: Nov 7 08:28:56 hockey in.tftpd[27601]: RRQ from 192.168.0.100 filename test.txt Let me know if that helps, or if I need to go into more detail. -- Jared Smith Community Relations Manager Digium, Inc.
Re: [asterisk-users] Snom 320 with TDM02B and echo problems
Jason White wrote: On Thu, Nov 08, 2007 at 10:22:41AM +0100, voip crazy wrote: I instaled an asterisk 1.2.X, with two lines FXO and two snom 3200 phones, on an amd_64 processor. All goes well, the voice is clear on the remote side but in the Voip side, where the Snom 320 is placed, I hear my voice, but don't in the line, the echo is on the phone. I just play with zapata gain values and with the Snom mic volume, but the echos does not disapperars. the phone is updated to firmware 6.5.12, the last i have found. Mine came with 7.1.8. Perhaps you should contact Snom to find out whether you can obtain the new version of the firmware. http://www.snom.com/en/no_cache/firmware.html http://wiki.snom.com/Main_Page The 7.x versions can be installed on all snom3x0 but they are in beta state for anything except the 370. The wiki describes how to do that. Not sure if upgrading helps with your problem though. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] If caller id is null set to a specific number
Doug Lytle wrote:
Jon Weisman wrote:
All,
If someone calls into my asterisk box and has a private number I would
like to set the callers id to a specific telephone number, only when
the ANI is missing, otherwise if present just pass it along. Any ideas?
[incoming]
exten = s,1,Gosubif($[${CALLERID(number)} = ]?set-cid,s,1:2)
[set-cid]
exten = s,1,Set(CALLERID(number)=5551212)
exten = s,n,Return()
ANI is not Caller*ID. A caller can block their Caller*ID, but not their
ANI.
It is CALLERID(num), not CALLERID(number)
In 1.2+ you can do it as (all one line):
exten = s,1,ExecIf($[${CALLERID(num)} = ],Set,CALLERID(num)=401212)
OR
exten = s,1,ExecIf($[${LEN(${CALLERID(num)})} =
0],Set,CALLERID(num)=401212)
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Re: [asterisk-users] Snom 320 with TDM02B and echo problems
I have found the new 7.x.x series firmware to be pretty much unusable in speakerphone mode, which is slightly disappointing as I like the Snom phones. PaulH On Fri, 2007-11-09 at 03:34 +0100, Philipp Kempgen wrote: Jason White wrote: On Thu, Nov 08, 2007 at 10:22:41AM +0100, voip crazy wrote: I instaled an asterisk 1.2.X, with two lines FXO and two snom 3200 phones, on an amd_64 processor. All goes well, the voice is clear on the remote side but in the Voip side, where the Snom 320 is placed, I hear my voice, but don't in the line, the echo is on the phone. I just play with zapata gain values and with the Snom mic volume, but the echos does not disapperars. the phone is updated to firmware 6.5.12, the last i have found. Mine came with 7.1.8. Perhaps you should contact Snom to find out whether you can obtain the new version of the firmware. http://www.snom.com/en/no_cache/firmware.html http://wiki.snom.com/Main_Page The 7.x versions can be installed on all snom3x0 but they are in beta state for anything except the 370. The wiki describes how to do that. Not sure if upgrading helps with your problem though. Regards, Philipp Kempgen ___ --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] Snom 320 with TDM02B and echo problems
I have had similar problems. My solution was to upgrade to a sangoma a200d that has echo canellation built in. I will NEVER buy an fxo card that doesn't have onboard echo cancellation ever again. There is just no other way to get good sound and no echo. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Friday, November 09, 2007 12:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Snom 320 with TDM02B and echo problems I have found the new 7.x.x series firmware to be pretty much unusable in speakerphone mode, which is slightly disappointing as I like the Snom phones. PaulH On Fri, 2007-11-09 at 03:34 +0100, Philipp Kempgen wrote: Jason White wrote: On Thu, Nov 08, 2007 at 10:22:41AM +0100, voip crazy wrote: I instaled an asterisk 1.2.X, with two lines FXO and two snom 3200 phones, on an amd_64 processor. All goes well, the voice is clear on the remote side but in the Voip side, where the Snom 320 is placed, I hear my voice, but don't in the line, the echo is on the phone. I just play with zapata gain values and with the Snom mic volume, but the echos does not disapperars. the phone is updated to firmware 6.5.12, the last i have found. Mine came with 7.1.8. Perhaps you should contact Snom to find out whether you can obtain the new version of the firmware. http://www.snom.com/en/no_cache/firmware.html http://wiki.snom.com/Main_Page The 7.x versions can be installed on all snom3x0 but they are in beta state for anything except the 370. The wiki describes how to do that. Not sure if upgrading helps with your problem though. Regards, Philipp Kempgen ___ --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, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] Kernel Native PCIE Network Cards?
Hi, I am getting a new sangoma t1 card soon and that will max out my slots, which means i need to take out a card. I am going to take out my pci network interface card (10/100) I have an open pci-e slot i have never used in the machine so i am going to buy a pci-e 10/100 or gigabit network adapter. I want to find one that works natively with the linux kernel. I hate using hardware that requires additional drivers in linux and have read tons of nightmares of people trying to get pci-express nic drivers to work with linux. So if someone could point me to a card that is natively supported in 2.6.15 i would appreciate it. Thanks Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] Your favorite desktop wifi sip hardphone ?
Hi, Which is your favorite desktop wifi sip hardphone ? I'm looking for something like http://www.mitel.com/DocController?documentId=19401 which could be easily moved from one meeting room to another. (In this specific case, finding an electrical plug to power a large desktop phone is seen more relevant than finding an PoE Ethernet plug or using a mobile handset.) Which product would you recommend ? 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
