[asterisk-users] caller id issue for INDIA
hi all, how to set the caller id facility for the TDM400p card in INDIA. thanks sandeep.s ___ -- 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] VoIP Users Conference today at 1PM Friday EST
http://VoipUsersConference.org for how to join us live IRC freenode.net #voip-users-conference Question: have any US presidential candidates said anything about technologies of interest to us? I hear Obama plans to have a tech cabinet post, so maybe we won't hear about the tubes anymore. Junction Networks CEO Michael Oeth is our guest today to talk about their onSIP.com hosted stuff. I've tested it and I know Junction works well because I've used it for a couple of years now. The conference starts one hour later today, 1 PM EST (10 AM Pacific, 11 Mountain, 12 Noon Central, 6 PM UK/GMT) ___ -- 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] caller id issue for INDIA
Hi, For the caller id there is a patch available for digium cards. you can patch that file. I am not aware about those files. so please refer some googleing. On Jan 18, 2008 2:57 PM, sandeep [EMAIL PROTECTED] wrote: hi all, how to set the caller id facility for the TDM400p card in INDIA. thanks sandeep.s ___ -- 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 -- Thank you with regards, Gopal, ___ -- 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
Kevin Kiely wrote: Great suggestion, thanks. The boot failed with the mac-phone.cfg removed. I re-touched the file and followed your suggestion. Any way of removing the call forwarding feature via the xml configs? Kevin Kiely wrote: I have a remote user on a Polycom IP Phone who has set call forwarding by accident and is away from the phone. Does anyone know of a way to remotely un-forward the phone? I tried to reboot the phone but that didn't work and removing the mac-phone.cfg caused problems Remove the XML element tag from within mac-phone.cfg that it updated with the forwarding information and then reboot it again. I know there's a way to disable DND on the polycom's via sip.cfg. I'm not sure about call forward. I would need to check the master config file. -- -- 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
[asterisk-users] Upgrading to Asterisk 1.4 :: Avoiding the hidden traps
In my series of articles about Asterisk 1.4, I've added a checklist for those of you upgrading to 1.4 from 1.2. As always, I appreciate feedback on important things I've forgotten... http://www.voip-forum.com/category/asterisk/asterisk14/ Have a nice weekend! /Olle ___ -- 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] SIP
Hello, For each incoming or outgoing call, sip hardphones I'm using, turn BLF on and off like this: the first call (after leaving idle status) turns 1st BLF on, the second one turns 2nd BLF and so on, when a call is hanged, its BLF is turn off. My first question is : do you think such behaviour is general ? My 2nd question is : using AMI, how can I tell for a given extension : 1. the number of ongoing calls (both incoming or outgoing ones), 2. classify them by time of creation (so that I can be somehow be certain to tie each call with each BLF) I've seen AMI ExtensionState Action ( http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+ExtensionState) but it won't tell the number of calls. To complete my request, I must add that when I'm requesting an extension ongoing calls listing, I haven't previously issued any request asking Asterisk to notify calls events for that extension. Any hint ? Cheers ___ -- 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] Automatic call-out problem
Hello!
My setup is Asterisk 1.2.26 with Zaptel 1.2.22.1, libpri-1.2.7 on
Fedora Core 4. I am making automatic call-out campaign with this setup
on 4 PRI. The scripts for this:
caller php script write this to outgoung folder:
fwrite($outfile,Channel: Zap/g1/$phonenumber\n);
fwrite($outfile,MaxRetries: 0\n);
fwrite($outfile,RetryTime: 5\n);
fwrite($outfile,WaitTime: 20\n);
fwrite($outfile,Context: 0100q\n);
fwrite($outfile,Callerid: $dbid\n);
fwrite($outfile,Extension: $phonenumber\n);
fwrite($outfile,Set: par_telszam=$phonenumber\n);
extensions.conf:
[0100q]
exten = _.,1,Wait(1)
exten = _.,n,Set(__TRIES=1)
exten = _.,n,Set(__FMT_DATE=%Y-%m-%d %H:%M:%S)
exten = _.,n,Set(__SZAM=${par_telszam})
exten = _.,n,System(echo -e
${SZAM}\,felvette\,${STRFTIME(${EPOCH},,${FMT_DATE})}
/tmp/0100q_0.txt)
exten = _.,n,Playback(0100q_0)
exten = _.,n,System(echo -e
${SZAM}\,99\,${STRFTIME(${EPOCH},,${FMT_DATE})}
/tmp/0100q_1v.txt)
exten = _.,n(valasztas),Set(TIMEOUT(response)=5)
exten = _.,n,Set(TIMEOUT(digit)=1)
exten = _.,n,Background(0100q_1)
exten = t,1,System(echo -e
${SZAM}\,timeout\,${STRFTIME(${EPOCH},,${FMT_DATE})}
/tmp/0100q_1.txt)
exten = t,n,Goto(0100q_2,999,1)
exten = i,1,System(echo -e
${SZAM}\,invalid\,${STRFTIME(${EPOCH},,${FMT_DATE})}
/tmp/0100q_1.txt)
exten = i,n,Goto(0100q_2,999,1)
exten = 1,1,System(echo -e
${SZAM}\,1\,${STRFTIME(${EPOCH},,${FMT_DATE})} /tmp/0100q_1.txt)
exten = 1,n,Goto(0100q_2,999,1)
exten = 2,1,System(echo -e
${SZAM}\,2\,${STRFTIME(${EPOCH},,${FMT_DATE})} /tmp/0100q_1.txt)
exten = 2,n,Goto(0100q_2,999,1)
exten = 3,1,System(echo -e
${SZAM}\,3\,${STRFTIME(${EPOCH},,${FMT_DATE})} /tmp/0100q_1.txt)
exten = 3,n,Goto(0100q_2,999,1)
exten = 9,1,System(echo -e
${SZAM}\,9\,${STRFTIME(${EPOCH},,${FMT_DATE})} /tmp/0100q_1v.txt)
exten = 9,n,GotoIf($[${TRIES} = 4.00]?0100q_2,999,1)
exten = 9,n,Set(__TRIES=${MATH(${TRIES}+1)})
exten = 9,n,Wait(1)
exten = 9,n,Goto(_.,valasztas)
[0100q_2]
exten = 999,1,Wait(1)
exten = 999,n,Background(0100q_2)
exten = t,1,System(echo -e
${SZAM}\,timeout\,${STRFTIME(${EPOCH},,${FMT_DATE})}
/tmp/0100q_2.txt)
exten = t,n,Goto(0100q_9,999,1)
exten = i,1,System(echo -e
${SZAM}\,invalid\,${STRFTIME(${EPOCH},,${FMT_DATE})}
/tmp/0100q_2.txt)
exten = i,n,Goto(0100q_9,999,1)
exten = 1,1,System(echo -e
${SZAM}\,1\,${STRFTIME(${EPOCH},,${FMT_DATE})} /tmp/0100q_2.txt)
exten = 1,n,Goto(0100q_9,999,1)
[0100q_9]
exten = 999,1,Wait(1)
exten = 999,n,System(echo -e
${SZAM}\,elkoszont\,${STRFTIME(${EPOCH},,${FMT_DATE})}
/tmp/0100q_9.txt)
exten = 999,n,Playback(0100q_9)
exten = 999,n,Hangup
stats:
wc -l 0100q_0.txt = 14628
cut -d , -f 1 0100q_0.txt | sort | uniq -c -d | wc -l = 74
wc -l 0100q_1v.txt = 14300
cut -d , -f 1 0100q_1v.txt | sort | uniq -c -d | wc -l = 498
grep ,99, 0100q_1v.txt | cut -d , -f 1 | sort | uniq -c -d | wc -l = 66
cut -d , -f 1 0100q_1.txt | sort | uniq -c -d | wc -l = 0
same for 2 and 9
Txt format is number,string,date.
Caller script call every number once if call was successful. I
checked. Therefore there can not be duplicates in _0, there can not be
multiple 99 string for a number. Looks like there is some variable
problem but I did not find where is it. Because there is thousands of
successful calls the script should be correct I think.
Any idea why is it happen? Is it a bug or I am just blind?
bye,
a
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Re: [asterisk-users] Using MysqlPool Application 1.4
can we use it for cdr information too ? Tilghman Lesher wrote: On Tuesday 18 December 2007 03:59:04 Cyril SCETBON wrote: Is anyone in the same troubles ? Do you advice me another solution to connect to my database ? See func_odbc.conf. -- Cyril SCETBON ___ -- 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] Two Asterisks behind NAT and need to link them using IAX trunk
Hi; Via OpenVPN or port forwarding is known for me, but via SSH is new for me, how I can do it and what is the difference by SSH and OpenVPN? Regards Bilal - Good question. I have never tried tunneling IAX over SSH but it seems like it should work just like anything else. How about a port opened up for OpenVPN. You know you can run IAX on any port you wish, port 80 may work for you if you have some extra external IPs not being used for HTTP. The same is true for OpenVPN. Thanks, Steve Totaro On Jan 17, 2008 8:09 PM, John Constalgie [EMAIL PROTECTED] wrote: Hi there this is an interesting topic that I see here and a problem that I am trying to solve too. But I was wondering if the forwarding solution will work for my case. So I have two Asterisk boxes A and B. A is behind a corporate NAT such that A can SSH to B, but not vice versa( One-way SSH ) . The UDP port 5060 of the corporate NAT is blocked off and I will not be able to have it unblocked for security reasons. Hence, is my only choice using an SSH tunnel between A and B for the IAX connection to work? Will it work though with that One-way SSH factor mentioned before? Thanks John Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ -- 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] Advice on AMI and SIP (was: SIP)
Hello, For each incoming or outgoing call, sip hardphones I'm using, turn BLF on and off like this: the first call (after leaving idle status) turns 1st BLF on, the second one turns 2nd BLF and so on, when a call is hanged, its BLF is turn off. My first question is : do you think such behaviour is general ? My 2nd question is : using AMI, how can I tell for a given extension : 1. the number of ongoing calls (both incoming or outgoing ones), 2. classify them by time of creation (so that I can be somehow be certain to tie each call with each BLF) I've seen AMI ExtensionState Action (http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+ExtensionState) but it won't tell the number of calls. To complete my request, I must add that when I'm requesting an extension ongoing calls listing, I haven't previously issued any request asking Asterisk to notify calls events for that extension. Any hint ? Cheers ___ -- 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 ip phne 7911G with asterisk
I'm running Asterisk 1.4.17 and as far as I know it only happens on the 7911g. And it's only issue when a user from a 7911g phone is leaving a message. Calls between sip users and PSTN sound good. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Pinedo Zamalloa Sent: Friday, January 18, 2008 8:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cisco ip phne 7911G with asterisk On Wed, Jan 16, 2008 at 10:26:04AM -0500, Anciso, Roy wrote: Now that you have your 7911g phone up running, would you mind checking the audio quality when leaving a voicemail for on another local asterisk user from this phone? I have a 7911g and I hear loud audio taps from the phone. The 7961g phone doesn't have this issue. I'm just trying to rule out the phone. Thanks I would try if a I have more time. Nowadays I have problems with the sound quality in the conversatio. 7941g with SIP-8-0-3 firwmware sounds well but 7911G with the same firmware version sounds really bad in an Asterisk 1.2.15. I listen echo and noise taps when I talk with other sip users of the same Asterisk. I have test the phone with the same firmware againts the lastest version of Asterisk (subversion stable tree) and there is no problem with the sound quality. Do you know what is the problem in this case? Or if there is a bug in Asterisk 1.2 that could affect in that way the audio quality? Thanks, However againts the 1.4.17 version there is no problem -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Pinedo Zamalloa Sent: Wednesday, January 16, 2008 10:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cisco ip phne 7911G with asterisk On Tue, Jan 15, 2008 at 01:14:42PM +, Christian Pinedo wrote: hi, I'm trying to configure a Cisco IP Phone 7911G in order to work with Asterisk. I have loaded the 8.3.3 SIP Firmware of Cisco through a DHCP and a TFTP server. All seems ok but a file that is downloaded : term06.default.loads (I understand that is for 7906 model) instead of term11.default.loads (I understand that is for 7911 model). In any case the phone reboots well. At this moment I thought that the phone should ask the SEPmac.xml.cnf file but it asks CTLSEPmac.tlv all the time. I don't have this file in the server and it tries to download every few seconds whitout asking another file. According to what I have read this file shouldn't be neccesary and, when the phone cann't obtain it, the phone should ask SEPmac.xml.cnf. I don't know if I'm doing something bad or if it could be a issue of the firmware version. I would thank some clue. Thanks, It was a TFTP server issue. The classical TFTP server used in the unix world responds to queries with bad error codes. I finally used aTFTPD that does this well so the phone understands that there's no CTLSEP file and then asks for SEP file. -- Christian Pinedo Zamalloa (zako) PGP key at: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x828D0C80 Fingerprint: 7BFF 4105 F46B 7977 BD96 348C 1007 4FF8 828D 0C80 ___ -- 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 -- Christian Pinedo Zamalloa (zako) PGP key at: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x828D0C80 Fingerprint: 7BFF 4105 F46B 7977 BD96 348C 1007 4FF8 828D 0C80 ___ -- 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] Zaptel timing on TE405P
On 1/17/08, Atis Lezdins [EMAIL PROTECTED] wrote: On 1/17/08, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Jan 17, 2008 at 03:09:59PM +0200, Atis Lezdins wrote: Hi, I'm wondering why zttest shows Best: 99.976 -- Worst: 99.967 -- Average: 99.971469, Difference: 99.971469 Shouldn't it be 100% as timing is hardware and comes from PRI? Am I missing some kernel config? It may be slightly different. Your system clock may be slightly off. But more importantly, zttest doesn't start and stop messuring time at exactly the right spot. Anything i can improve? I think - zttest should do it correctly, as manpage says - definite pass is 100% or 99.99% I'm just having some issues with faxing, so i thought this could be a problem. Ping. Any ideas what i could do to improve timing accuracy? Some kernel options? Newer kernel? Currently I have kernel from RPM: Linux asterisk2 2.6.22.7-57.fc6 #1 SMP Fri Sep 21 19:45:12 EDT 2007 x86_64 x86_64 x86_64 GNU/Linux Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- 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] Two Asterisks behind NAT and need to link themusing IAX trunk
It is possible to run openVPN in TCP mode over an SSH tunnel. Don't turn compression on on both though - I'd just switch it on the openVPN if you have to. You will probably find the speech is rather choppy due to the delays and fragmentation, but I have done this. Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: 18 January 2008 12:21 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Two Asterisks behind NAT and need to link themusing IAX trunk Hi; Via OpenVPN or port forwarding is known for me, but via SSH is new for me, how I can do it and what is the difference by SSH and OpenVPN? Regards Bilal - Good question. I have never tried tunneling IAX over SSH but it seems like it should work just like anything else. How about a port opened up for OpenVPN. You know you can run IAX on any port you wish, port 80 may work for you if you have some extra external IPs not being used for HTTP. The same is true for OpenVPN. Thanks, Steve Totaro On Jan 17, 2008 8:09 PM, John Constalgie [EMAIL PROTECTED] wrote: Hi there this is an interesting topic that I see here and a problem that I am trying to solve too. But I was wondering if the forwarding solution will work for my case. So I have two Asterisk boxes A and B. A is behind a corporate NAT such that A can SSH to B, but not vice versa( One-way SSH ) . The UDP port 5060 of the corporate NAT is blocked off and I will not be able to have it unblocked for security reasons. Hence, is my only choice using an SSH tunnel between A and B for the IAX connection to work? Will it work though with that One-way SSH factor mentioned before? Thanks John Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ -- 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 and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ -- 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] Channels ID / Soft Hang Up
Hello, I am wanting to close a specific channel for example; SofthangUp(SIP/EXTEN-UNIQUEID) but the problem is the channel is assigned a unique id as well. The need fits into the idea of receiving a call from a higher status user and thus closing a specific channel to allow the higher priority call to route through the dial plan to the freed extension. Any ideas welcome. Many thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 17 January 2008 01:54 To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 42, Issue 58 *** WARNING *** This mail has originated outside your organization, either from an external partner or the Global Internet. Keep this in mind if you answer this message. Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. Re: [IAX] Up-to-date list of soft- and hardphones? (Gordon Henderson) 2. Re: Can DB() use SQLite instead of BerkeleyDB? (Tilghman Lesher) 3. Re: WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101' (Andrew Joakimsen) 4. Re: Digium Part#'s (Was: Difference between TE121 and TE122) (Kevin P. Fleming) 5. asterisk to mysql database! (Naveen Palani) 6. Re: asterisk to mysql database! (Simon Elliston Ball) 7. Asterisk 1.4.17 and RXFAX via T38 (Robert Moskowitz) 8. Re: Unable to open master device '/dev/zap/ctl' (Chris Bagnall) 9. Re: [IAX] Up-to-date list of soft- and hardphones? (Vincent) 10. Re: Can DB() use SQLite instead of BerkeleyDB? (Vincent) 11. Re: asterisk to mysql database! (Tilghman Lesher) 12. Re: [IAX] Up-to-date list of soft- and hardphones? (Tim H. Panton) 13. HDLC errors (Steven) 14. Re: HDLC errors (Russell Bryant) 15. AddQueueMember and Flash Operator Panel ([EMAIL PROTECTED]) 16. Re: HDLC errors (Steve Totaro) 17. Anyone Using a Dell PowerEdge T105 in Production (Steve Totaro) 18. Problem with a channel (Ruben Zamora) 19. Re: HDLC errors (Andrew Joakimsen) 20. IMAP client in asterisk not trying to contact IMAPserver (KodaK) 21. Asterisk Now Beta 6 and CISCO IP 7910 ([EMAIL PROTECTED]) 22. Re: Anyone Using a Dell PowerEdge T105 in Production (Erik Anderson) 23. Re: Anyone Using a Dell PowerEdge T105 in Production (Steve Totaro) 24. Asterisk on ClarkConnect (shadowym) 25. Re: Unable to open master device '/dev/zap/ctl' (Walter Willis) 26. Re: Anyone Using a Dell PowerEdge T105 in Production (Erik Anderson) -- Message: 1 Date: Wed, 16 Jan 2008 18:08:23 + (GMT) From: Gordon Henderson [EMAIL PROTECTED] Subject: Re: [asterisk-users] [IAX] Up-to-date list of soft- and hardphones? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed On Wed, 16 Jan 2008, Vincent wrote: Hello There's a lot of information on VoIP at www.voip-info.org ... but there's also a lot of outdated information there as well :-/ Since SIP is a pain to use when NAT is involved, especially when both the Asterisk server and the remote phones are behind NAT... I'd like to try IAX to see how it works and if it solves the issue. I'd like to start with a softphone (Windows only), and then, if tests prove successfully, buy a hardphone. What would be your recommendations? IDEFISK or Zoiper as it's called now. However, you'll need to do similar things to your asterisk box router if it's behind NAT for IAX as you do for SIP. (You will need a static IP address on the NAT router and port-forward 4569 to the asterisk box, just as you'd port-forward 5060 and 1-2 for SIP) And a SIP phone behind a NAT router is also solvable if it supports STUN. I know that SIP behind NAT isn't perfect, but with care, it's very usable and workable. I have many installations doing just this, as I'm sure many others on the list have too. Gordon -- Message: 2 Date: Wed, 16 Jan 2008 12:10:35 -0600 From: Tilghman Lesher [EMAIL PROTECTED] Subject: Re: [asterisk-users] Can DB() use SQLite instead of BerkeleyDB? 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 On Wednesday 16 January 2008 10:02:12 Vincent wrote:
Re: [asterisk-users] Device state of SIP doesn't change
Atis Lezdins wrote: On 1/17/08, Mark Michelson [EMAIL PROTECTED] wrote: Atis Lezdins wrote: Hi, I'm wondering - why SIP device state doesn't get updated to anything else, except Not In Use. For queue call (with Local channel) i get: app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: The device state of this queue member, Agent/21168, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. Of course, i checked UPGRADE.txt, and lot of other resources, enabled few settings in sip.conf, but this still doesn't change. my sip.conf is: [general] port = 5060 bindaddr = 0.0.0.0 context = default-external tos_sip=0x18 tos_audio=0x18 callerid = Unknown dtmfmode=rfc2833 ignoreregexpire=yes limitonpeer=yes notifyringing=yes notifyhold=yes allowsubscribe=yes call-limit=1 and the corresponding realtime entry is: name: 21168 accountcode: NULL amaflags: NULL callgroup: NULL callerid: device 21168 canreinvite: no context: default-sip defaultip: NULL dtmfmode: rfc2833 fromuser: NULL fromdomain: NULL fullcontact: NULL host: dynamic insecure: NULL language: NULL mailbox: [EMAIL PROTECTED] md5secret: NULL nat: yes deny: NULL permit: NULL mask: NULL pickupgroup: NULL port: 5061 qualify: no restrictcid: NULL rtptimeout: NULL rtpholdtimeout: NULL secret: xxx type: friend username: 21168 disallow: allow: all musiconhold: NULL regseconds: 1200593168 ipaddr: xxx.xxx.xxx.xxx regexten: cancallforward: yes setvar: Any help would be appreciated. Regards, Atis The relevant portion of UPGRADE.txt mentions that a call-limit is necessary in order for SIP devices to report proper device state. I see in your sip.conf file that you have set call-limit in the general section. This setting, however, may only be set per peer (or user). Unfortunately, there's no warning message output if an unrecognized option is set in the general section. Mark, thanks for pointing this out. However, i was stuck without any success, until i tried adding my phone in static config - then it magically worked. So, i could use rtcachefriends=yes but that's something i would really like to avoid. Is this considered a bug? There's nothing in docs saying that state information is incompatible with Realtime. Regards, Atis After further discussion regarding this in #asterisk this morning, it would appear that communicating proper device state with realtime peers/users does not work properly. I would tentatively consider this a bug since I would expect that anything that works statically should also work in realtime as well. However, since I have not done a ton of work with chan_sip myself, there could be some subtle (or not so subtle) reason why this was purposely not implemented. Sorry I can't be more authoritative on this matter. Mark Michelson ___ -- 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] OT: Call for beta testers (well... perhaps late Alpha).
We've just launched the beta of a free service which is, really, still only JUST out of the alpha stages. http://www.voipmagnet.com The basic idea is this: it's an opt-in directory focused on VoIP contact info (with elements of social networking and privacy control). Again, the service is very rough, but we'd like input from the VoIP community. There are a good many things that are likely buggy, broken, or not yet implemented, but we feel that's what a beta is for. If you have any questions/concerns/issues/feedback/abuse, feel free to send it to me directly via email or post it to the list. Some things we know will be changed soon: -ability to add multiple VoIP accounts with each login (this is in the DB, but the interface elements are there yet -- we didn't like the way they looked first pass 'round) -ability to invite friends to join up so you can share contact info with friend groups without making everything (or anything) public -search optimisations (searching is functional, but a bit rough. We're looking for any and all input there, of course) We've a whole plate load of ideas for what to cram into the service, but the intent is to keep it free and provider-agnostic, but still maintain a centralised location where anyone can go and look up friends or coworkers to find out what VoIP service they're using and how to get in touch. We're open to any and all suggestions to what to add/change/fix to make this a service out of which the community will get some use. -- Neil Fusillo CEO Infinideas, inc. http://www.infinideas.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] R2-Unicall Asterisk as CPE and as CO
Hi! Im having some troubles trying to configure * as a bridge between a telco and a pbx with R2, the scenario is this: E1/R2-E1/R2 | Telco |-| * |-| PBX| | (Telmex) | - | | I can receive calls from the telco and can place calls to the pbx, I also can place calls to the telco.. but I can't receive any calls from the pbx. When receive a call from the pbx I get this: cause 32771 - T3 timed out If I connect the pbx directly to the telco there is no problem, the calls are stablished correctly. Im using the package at: http://www.moythreads.com/astunicall/downloads/ http://www.moythreads.com/astunicall/files/astunicall-1.2.25-0.1.tar.gz that contains: asterisk-1.2.25 spandsp-0.0.4 unicall-0.0.5pre1 libmfcr2-0.0.3 libsupertone-0.0.2 libunicall-0.0.3 zaptel-1.2.22 My zaptel.conf is this: loadzone=mx defaultzone=mx span=1,1,0,cas,hdb3 span=2,1,0,cas,hdb3 span=3,0,0,cas,hdb3 span=4,0,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 cas=32-46:1101 cas=48-62:1101 cas=63-77:1101 cas=79-93:1101 cas=94-103:1101 cas=110-124:1101 and unicall.conf is this: [channels] usecallerid=no hidecallerid=no callwaitingcallerid=no threewaycalling=no transfer=no cancallforward=no callreturn=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=yes loglevel=255 protocolclass=mfcr2 protocolvariant=mx,10,4,16 group=1 protocolend=cpe context=incoming1 channel = 1-15 channel = 17-31 group=2 protocolend=cpe context=incoming2 channel = 32-46 channel = 48-62 protocolvariant=mx,10,8 group=3 immediate=yes usecallerid=yes protocolend=co context=incoming3 channel = 63-77 channel = 79-93 group=4 protocolend=co context=incoming4 channel = 94-103 channel = 110-124 The port #1 of a TE405P card is connected to the telco and the port #3 is connected to the pbx. I've changed the line (chan_unicall.c): uc_callparm_calling_party_category(callparms, UC_CALLER_CATEGORY_NATIONAL_SUBSCRIBER_CALL); to uc_callparm_calling_party_category(callparms, UC_CALLER_CATEGORY_NATIONAL_PRIORITY_SUBSCRIBER_CALL); because without this I cant receive calls from the telco. With or without this I can't place calls to the pbx. When I receive a call from the telco I place it directly to the pbx.. and that works ok: Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 - 0001 [1/IDLE/Idle /Idle ] Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 Detected Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 Creating a new call with CRN 32770 Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 1101 - [2/DETECTED/Seize ack /Seize ack] Jan 16 12:27:01 NOTICE[4136] chan_unicall.c: Unicall/2 event Detected Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 - 4 on [2/DETECTED/Seize ack /Seize ack] Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 6 on - [2/DETECTED/Group C /Category req ] Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 - 4 off [2/DETECTED/Group C /Category req ] Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 6 off - [2/DETECTED/Group C /Category req ] Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 - 2 on [2/DETECTED/Group C /Category req ] Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 1 on - [2/DETECTED/Group C /ANI request ] Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 - 2 off [2/DETECTED/Group C /ANI request ] Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 1 off - [2/DETECTED/Group C /ANI request ] Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 - F on [2/DETECTED/Group C /ANI request ] Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 5 on - [2/DETECTED/Group A /DNIS request ] Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 - F off [2/DETECTED/Group A /DNIS request ] Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 5 off - [2/DETECTED/Group A /DNIS request ] Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 - 6 on [2/DETECTED/Group A /DNIS request ] Jan 16 12:27:01
Re: [asterisk-users] asterisk-1.2.26.tar.gz Thoughts?
1.2.26 Works a treat here on several 10s of sites - We are just now starting to look at 1.4.x as it seems that is is begining to stabilise. Regards, Steve On 1/18/08, Matt [EMAIL PROTECTED] wrote: ** Bump ** On Jan 17, 2008 3:00 PM, Matt [EMAIL PROTECTED] wrote: What are people's thoughts on asterisk 1.2.26? Any show stopping bugs? 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] Asterisk and postgresql query
Hi all I need some help with query on Postresql Database. I have some tables on a DB, and I wanna to retrieve status on field in this tables... I explain: ON the DB table, I have this field: = Service Order = the situation about Product Order Now, when a cliente call to phone pluged on Asterisk, he follow the URA, by the system. So, the URA pass instruction to client. to him press the keys, informed the Service Order, maybe 1234. In this case, the system will be compare the sequence order, and take the date from system, i.e., from machine where Asterisk is run. Well, the system take this to field's: date and number of Service Order. With this numbers, I want the system check this information on a PostgreSQL DB, via some ODBC query or whatever, and bring the status for Service Order. I know that this can be appears some strange! And to be add, my English is poor and horrible! Some one can help? Thanks for any help... -- 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] asterisk-1.2.26.tar.gz Thoughts?
** Bump ** On Jan 17, 2008 3:00 PM, Matt [EMAIL PROTECTED] wrote: What are people's thoughts on asterisk 1.2.26? Any show stopping bugs? ___ -- 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] OT: To Admins: Missing DNS for list server
Hi! A little OT, but I have a question about the list server. Is it okay for the list server 216.207.245.17 to have no DNS name? [EMAIL PROTECTED]:~$ host 216.207.245.17 Host 17.245.207.216.in-addr.arpa. not found: 3(NXDOMAIN) I tried different DNS. Mailserver may not accept mails from hosts without a valid hostname (and some even without a valid reverse DNS hostname). So it may be good to have one. Sorry for the noise. Shade and sweet water! Stephan -- | Stephan SeitzE-Mail: [EMAIL PROTECTED] | | PGP Public Keys: http://fsing.rootsland.net/~stse/pgp.html | signature.asc Description: Digital signature ___ -- 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] Device state of SIP doesn't change
Mark Michelson wrote: Atis Lezdins wrote: On 1/17/08, Mark Michelson [EMAIL PROTECTED] wrote: Atis Lezdins wrote: Hi, I'm wondering - why SIP device state doesn't get updated to anything else, except Not In Use. For queue call (with Local channel) i get: app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: The device state of this queue member, Agent/21168, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. Of course, i checked UPGRADE.txt, and lot of other resources, enabled few settings in sip.conf, but this still doesn't change. my sip.conf is: [general] port = 5060 bindaddr = 0.0.0.0 context = default-external tos_sip=0x18 tos_audio=0x18 callerid = Unknown dtmfmode=rfc2833 ignoreregexpire=yes limitonpeer=yes notifyringing=yes notifyhold=yes allowsubscribe=yes call-limit=1 and the corresponding realtime entry is: name: 21168 accountcode: NULL amaflags: NULL callgroup: NULL callerid: device 21168 canreinvite: no context: default-sip defaultip: NULL dtmfmode: rfc2833 fromuser: NULL fromdomain: NULL fullcontact: NULL host: dynamic insecure: NULL language: NULL mailbox: [EMAIL PROTECTED] md5secret: NULL nat: yes deny: NULL permit: NULL mask: NULL pickupgroup: NULL port: 5061 qualify: no restrictcid: NULL rtptimeout: NULL rtpholdtimeout: NULL secret: xxx type: friend username: 21168 disallow: allow: all musiconhold: NULL regseconds: 1200593168 ipaddr: xxx.xxx.xxx.xxx regexten: cancallforward: yes setvar: Any help would be appreciated. Regards, Atis The relevant portion of UPGRADE.txt mentions that a call-limit is necessary in order for SIP devices to report proper device state. I see in your sip.conf file that you have set call-limit in the general section. This setting, however, may only be set per peer (or user). Unfortunately, there's no warning message output if an unrecognized option is set in the general section. Mark, thanks for pointing this out. However, i was stuck without any success, until i tried adding my phone in static config - then it magically worked. So, i could use rtcachefriends=yes but that's something i would really like to avoid. Is this considered a bug? There's nothing in docs saying that state information is incompatible with Realtime. Regards, Atis After further discussion regarding this in #asterisk this morning, it would appear that communicating proper device state with realtime peers/users does not work properly. I would tentatively consider this a bug since I would expect that anything that works statically should also work in realtime as well. However, since I have not done a ton of work with chan_sip myself, there could be some subtle (or not so subtle) reason why this was purposely not implemented. Sorry I can't be more authoritative on this matter. Mark Michelson After some discussion on IRC, and reviewing my initial reply to you, I should clarify that proper device state reporting for realtime SIP peers does work with rtcachefriends enabled. I believe I will start up a branch soon to work out the details of getting proper device state reported for realtime SIP peers which are not cached. Mark Michelson ___ -- 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] HDLC errors
Well how do you test with one of the cheap continuity testers or with a Fluke or similar? Also just for the heck of it... I assume in your zapata.conf your lbo is set to 0 try 1 see if it makes a difference. Most of the time your CSU/DSU/NIU will autodetect this change. But just try it see what happens and if possible post back with what equipment the telco installed for the T1. On Jan 17, 2008 7:49 PM, Steven Kurylo [EMAIL PROTECTED] wrote: You mention went into production, Did this imply moving of the system from a testing room into a server-location? Other (longer) cables? Unplugged the current system and hooked up a new, longer, cable to the asterisk system. The cable is RJ48 STP, about 100 feet. However we ran several cables and swapping them around doesn't make a difference; they all test good too. We could be having bad luck with them :-) I was thinking of moving the server to be beside the telco box, but that is a large undertaking. Perhaps you can check with your telco wether they receive bad frames coming from you I gave them a call and they'll run a report and get back to me. Hopefully the patlooptest tonight will point to the problem. Thank you everyone for all your suggestions. ___ -- 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] Two Asterisks behind NAT and need to link them using IAX trunk
On Thu, Jan 17, 2008 at 11:06:22PM -0500, Steve Totaro wrote: Good question. I have never tried tunneling IAX over SSH but it seems like it should work just like anything else. SSH tunnels TCP alone. IAX is UDP. You can use it to create some sort of full-fledged VPN connection, but it is not trivial. Instead, you should probably go for openvpn. SSH is on top of TCP, so there is an inherent potential delay. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] caller id issue for INDIA
On Fri, Jan 18, 2008 at 02:57:23PM +0530, sandeep wrote: hi all, how to set the caller id facility for the TDM400p card in INDIA. http://bugs.digium.com/6683 Hmmm looks like it needs some love and care. I wasn't following it carefully. Can anybody update me on it? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] Accessing a MySQL database and using the same db for cdr
On Fri, Jan 18, 2008 at 03:33:07PM +0100, Cyril SCETBON wrote: Hi guys, Does someone use a mysql database for accessing data and in the same time for storing cdr ? if that is the case, which module is used ? thanks On a lightly used system with asterisk, I use ODBC to connect to MySQL for CDR and SIP/IAX/voicemail etc realtime storage. Other light application usage of the MySQL is also possible. -- Ron Wellsted [EMAIL PROTECTED] http://www.wellsted.org.uk N 52.567623, W 2.136111 Linux Counter No. 202120 Ekiga: 645022 signature.asc Description: Digital signature ___ -- 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] Accessing a MySQL database and using the same db for cdr
Hi guys, Does someone use a mysql database for accessing data and in the same time for storing cdr ? if that is the case, which module is used ? thanks -- Cyril SCETBON ___ -- 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] Device state of SIP doesn't change
On 1/17/08, Mark Michelson [EMAIL PROTECTED] wrote: Atis Lezdins wrote: Hi, I'm wondering - why SIP device state doesn't get updated to anything else, except Not In Use. For queue call (with Local channel) i get: app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: The device state of this queue member, Agent/21168, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. Of course, i checked UPGRADE.txt, and lot of other resources, enabled few settings in sip.conf, but this still doesn't change. my sip.conf is: [general] port = 5060 bindaddr = 0.0.0.0 context = default-external tos_sip=0x18 tos_audio=0x18 callerid = Unknown dtmfmode=rfc2833 ignoreregexpire=yes limitonpeer=yes notifyringing=yes notifyhold=yes allowsubscribe=yes call-limit=1 and the corresponding realtime entry is: name: 21168 accountcode: NULL amaflags: NULL callgroup: NULL callerid: device 21168 canreinvite: no context: default-sip defaultip: NULL dtmfmode: rfc2833 fromuser: NULL fromdomain: NULL fullcontact: NULL host: dynamic insecure: NULL language: NULL mailbox: [EMAIL PROTECTED] md5secret: NULL nat: yes deny: NULL permit: NULL mask: NULL pickupgroup: NULL port: 5061 qualify: no restrictcid: NULL rtptimeout: NULL rtpholdtimeout: NULL secret: xxx type: friend username: 21168 disallow: allow: all musiconhold: NULL regseconds: 1200593168 ipaddr: xxx.xxx.xxx.xxx regexten: cancallforward: yes setvar: Any help would be appreciated. Regards, Atis The relevant portion of UPGRADE.txt mentions that a call-limit is necessary in order for SIP devices to report proper device state. I see in your sip.conf file that you have set call-limit in the general section. This setting, however, may only be set per peer (or user). Unfortunately, there's no warning message output if an unrecognized option is set in the general section. Mark, thanks for pointing this out. However, i was stuck without any success, until i tried adding my phone in static config - then it magically worked. So, i could use rtcachefriends=yes but that's something i would really like to avoid. Is this considered a bug? There's nothing in docs saying that state information is incompatible with Realtime. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- 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-1.2.26.tar.gz Thoughts?
At 12:34 PM 1/18/2008, you wrote: Although for some of us, or at least me, no version of 1.4 has run for more than 72 hours before generating a kernel panic. I've tried about 6 versions, the early ones were good for about 10 minutes, the latest one lasted 3 days. Sadly I'm still stuck using the latest 1.2. Ira What type of Asterisk setup do you have? While my setup is not a large commercial setup I have seen asterisk 1.4 with a few calls going through it at once last for weeks if not months before it was restarted. Just curious. 1ghz Celeron, 1 gig ram, 120gb HD. An HP home desktop discarded by a client 2 year old Digium 4 FXO port card using only 3 ports and the Digium HP echo can 3 analog lines in 2 SIP lines in most outgoing via SIP Most incoming via analog phones are all Aastra 480i-CT Dial plan is hand written, likely a bit convoluted, but it's hard to avoid that. Seems like the panics were mostly to do with ZAP The internet runs over 192.168.0.XXX the phones run on 192.168.233.XXX The two networks are completely separate until they reach the router connected to the world. The only problem I have with the most current 1.2 is every month or three it thinks one of the phones has 5 active lines going and stops sending calls to it, restart gracefully and all is well again. Ira ___ -- 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] Two Asterisks behind NAT and need to link them using IAX trunk
Hi; How can I use SSH in that senario? Is there a link that can help to understand what I have to install and to configure? Regards Bilal -- bilal ghayyad wrote: Hi; Via OpenVPN or port forwarding is known for me, but via SSH is new for me, how I can do it and what is the difference by SSH and OpenVPN? SSH uses tcp. Openvpn, by default uses udp. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ -- 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] dtmf from Cell phones
when placing a call with a cellphone to the asterisk server, the dtmf recognition on asterisk is not working properly. It seems to duplicate the first digit so if I enter 123 on my cellphone, asterisk interprets it as 1123. I have messed around with relaxdtmf, toneduration. Calling from a landline or a sip phone works pefectly with no problems. I am using a digium tdm400 card with 2 fxo ports on asterisk 1.4.9. I tested the actual dtmf sound that asterisk is picking up by dialing from a cellphone and connecting to a sip phone attached to asterisk. when I hit a button I get a brief dtmf sound, then a pause and then the full dtmf sound. Anyone else seen this issue? Is this a bug or an incorrect configuration? ___ -- 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 from Cell phones
How are you getting the calls from the PSTN and into your Asterisk server? On Jan 18, 2008 5:41 PM, Dan Kirsche [EMAIL PROTECTED] wrote: when placing a call with a cellphone to the asterisk server, the dtmf recognition on asterisk is not working properly. It seems to duplicate the first digit so if I enter 123 on my cellphone, asterisk interprets it as 1123. I have messed around with relaxdtmf, toneduration. Calling from a landline or a sip phone works pefectly with no problems. I am using a digium tdm400 card with 2 fxo ports on asterisk 1.4.9. I tested the actual dtmf sound that asterisk is picking up by dialing from a cellphone and connecting to a sip phone attached to asterisk. when I hit a button I get a brief dtmf sound, then a pause and then the full dtmf sound. Anyone else seen this issue? Is this a bug or an incorrect configuration? ___ -- 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] AMIProxyPal - AMI Proxy Project
After having misunderstood some key elements of AstManProxy, I started to write my own proxy server for Asterisk AMI. I was under the impression that it required a mysql database to cache its data for some reason. (Is there another AMI proxy that uses a mysql database?) At any rate, I had written about 70% of the core functionality so I decided to continue on. I'm not a C programmer so having something in my preferred language to use and extend later on is nice. I still most of my development on Windows so I haven't had a chance to build any Linux binaries other than for debugging, but should have some ready in the next week or so as Linux testing continues. In the meantime, there are Win32 binaries in the repository. Currently I'm working on xml and ini based decorators to customize the packets to/from clients. I'm using the proxy for a re-write of an existing operator panel I have in order to make it cross platform, but I've released the proxy itself released under GPL. It is written in ObjectPascal using Lazarus IDE (0.9.24) with Freepascal compiler (2.2.0); Sources are available here: http://www.leebo.dreamhosters.com/AMIProxy/ -- Warm Regards, Lee The only thing that kept me out college...was high school. ___ -- 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] SAY TIME + PHPAGI + Timezone
On Friday 18 January 2008 14:13:55 Nitesh Divecha wrote: Is there any way to change the timezone on the fly? I have this little time clock program running on Asterisk system developed using PHPAGI. Currently, whenever user logs in, Asterisk will prompt the current system time using $agi-say_time(); which executes SAY TIME. Now the current timezone set on the system is PST, and I have a request to prompt multiple timezones based on the users location. Don't use SAY TIME. Use EXEC with the SayUnixTime application and the appropriate arguments. -- Tilghman ___ -- 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 1.6.0-beta1 released
The Asterisk.org development team has published Asterisk version 1.6.0-beta1. Everyone is encouraged to help test Asterisk 1.6, so that the release may be available soon. Asterisk 1.6 will be the first major release of Asterisk since 1.4, which was released just over one year ago. This release contains a number of new features, as well as architectural improvements for improved performance. A list of the new features is available in the CHANGES file: http://svn.digium.com/view/asterisk/tags/1.6.0-beta1/CHANGES?view=co Asterisk 1.6 also brings about a new release management style. This release management policies have been changed for Asterisk 1.6 to account for some of the things we have learned while maintaining Asterisk 1.2 and 1.4 in the past. For more information on the new release management policy, see the following thread on the asterisk-dev mailing list: http://lists.digium.com/pipermail/asterisk-dev/2007-October/030083.html The support levels for Asterisk 1.2 and 1.4 will not change in the near future. There are no current plans as to when the support of those releases will change. Those decisions will be made as a result of discussions in the developer community when the time comes, and a public announcement will be made with plenty of advance notice before anything changes. Thank you for the support, and we look forward to your feedback on this release! ___ -- 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] HDLC errors
Good to know they are helping you out. Post the outcome if you can, specifically if they send you the same model and revision card and if it resolves your issue. If you can try to make a note of your current firmware version before you send it off. On Jan 18, 2008 5:25 PM, Steven Kurylo [EMAIL PROTECTED] wrote: I have a suggestion. Have you contacted Digium technical support for assistance with resolving this issue? Excellent suggestion. Make sure you can give them SSH access and screen so you can see what they are doing. Before that, check (remake) your T1 cables and if it is punched down on a block, re-punch it. I'm used to vendors that aren't responsive, so I never even thought of it. They've told me to try running patlooptest (which I will tonight), to see if the problem is in the card. I received very good support from Digium. After running patlooptest they recommended that I return the card. During the loop tests I would sometimes pass a 60 second test. I very rarely passed a 300 second test on either span. I moved the card to the other slot and did some IRQ swapping as well to the same results. Thanks for all your suggestions. ___ -- 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] HDLC errors
I have a suggestion. Have you contacted Digium technical support for assistance with resolving this issue? Excellent suggestion. Make sure you can give them SSH access and screen so you can see what they are doing. Before that, check (remake) your T1 cables and if it is punched down on a block, re-punch it. I'm used to vendors that aren't responsive, so I never even thought of it. They've told me to try running patlooptest (which I will tonight), to see if the problem is in the card. I received very good support from Digium. After running patlooptest they recommended that I return the card. During the loop tests I would sometimes pass a 60 second test. I very rarely passed a 300 second test on either span. I moved the card to the other slot and did some IRQ swapping as well to the same results. Thanks for all your suggestions. ___ -- 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] Looking for business-grade SIP Softphone
Hi, I am looking for a good (not necessarily free) business-grade SIP Softphone that supports: 1) G729 2) Outlook contact integration (click on number to dial) 3) Remote provisioning (not a must, but a very nice to have) 4) Customizable skin (again, not a must but a nice to have) I've seen X-Lite (which has only 2 lines, not enough). The commercial version of X-Lite looks nice, but doesn't support provisioning. At the moment, it's my fallback plan. Regards, 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
[asterisk-users] Probably a simple question. Dial a call.
I would like to add a function to an existing application that will make an outgoing call. I found this example using the Manager API for originating a call to an extension. http://www.voip-info.org/wiki/index.php?page=Asterisk+manager+Example%3A+Originate I was wondering if the manager API was the correct way to do this and If anyone has any example code for doing this kind of thing? Thanks for the help. ___ -- 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] SAY TIME + PHPAGI + Timezone
of course, that assumes you're reading the variable in the AGI. SIP wrote: Use the Set(TZone=blah) command in the dialplan. I.e. Set(TZone=EST5EDT) N. Nitesh Divecha wrote: Hello All, Is there any way to change the timezone on the fly? I have this little time clock program running on Asterisk system developed using PHPAGI. Currently, whenever user logs in, Asterisk will prompt the current system time using $agi-say_time(); which executes SAY TIME. Now the current timezone set on the system is PST, and I have a request to prompt multiple timezones based on the users location. First part is easy to lookup from which area code the user is calling, now the second part is to set the timezone based on the area code and prompt the users correct time. For example, if 248 (MI) user dials into the system, then time clock has to prompt EST time and if 714 (CA) user dials in then prompt PST time. Any suggestions... Thanks Cheers, Nitesh ___ -- 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-1.2.26.tar.gz Thoughts?
At 11:53 AM 1/18/2008, you wrote: Apart from the fact asterisk 1.2 is in security maintenance mode only and wont get any other bugfixes it will be ok. Please consider using 1.4 as it's the official latest stable version. Although for some of us, or at least me, no version of 1.4 has run for more than 72 hours before generating a kernel panic. I've tried about 6 versions, the early ones were good for about 10 minutes, the latest one lasted 3 days. Sadly I'm still stuck using the latest 1.2. Ira ___ -- 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-1.2.26.tar.gz Thoughts?
On 15:00, Thu 17 Jan 08, Matt wrote: What are people's thoughts on asterisk 1.2.26? Any show stopping bugs? Apart from the fact asterisk 1.2 is in security maintenance mode only and wont get any other bugfixes it will be ok. Please consider using 1.4 as it's the official latest stable version. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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] Maximum retries/no reply to our critical packet
Hello All, Got one customer and he is getting disconnection within 15 seconds when he tries to make outbound calls. Initially, it was working fine without any glitches... Other customers on the same system are working fine, its just with this customer only. This is the error message thrown by Asterisk on the CLI: - Jan 18 12:23:30 WARNING[30532]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Response) Jan 18 12:23:30 WARNING[30532]: chan_sip.c:1245 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. Customer can receive inbound calls without any disconnections, its just when he tries to make outbound calls. All outbound calls are sent to Nextone SoftSwitch and default codec is G729a. Customer has Linksys SPA-2102 - firmware ver 3.3.6 and Asterisk version 1.2.18. Thanking in advance... Cheers, Nitesh ___ -- 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 MysqlPool Application 1.4
On Friday 18 January 2008 06:09:54 Cyril SCETBON wrote: Tilghman Lesher wrote: On Tuesday 18 December 2007 03:59:04 Cyril SCETBON wrote: Is anyone in the same troubles ? Do you advice me another solution to connect to my database ? See func_odbc.conf. can we use it for cdr information too ? You can use ODBC, sure, via cdr_odbc or (new in trunk) cdr_adaptive_odbc. -- Tilghman ___ -- 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] SAY TIME + PHPAGI + Timezone
Thanks everyone for the feedback... Manage to prompt time using EXEC with the SayUnixTime. Here is the snapshot of the timezone: - // Get current time $currentTime = time(); // Set the offset $offset = 3; // Modified time $modifiedTime = $currentTime + ($offset * 60 * 60); debug(Current time: $currentTime, 3); debug(Offset time: $modifiedTime, 3); // Say unix time $agi-exec(SayUnixTime, $modifiedTime,EST5EDT,ABdY \'digits/at\' IMp); Cheers, Nitesh Tilghman Lesher wrote: On Friday 18 January 2008 14:13:55 Nitesh Divecha wrote: Is there any way to change the timezone on the fly? I have this little time clock program running on Asterisk system developed using PHPAGI. Currently, whenever user logs in, Asterisk will prompt the current system time using $agi-say_time(); which executes SAY TIME. Now the current timezone set on the system is PST, and I have a request to prompt multiple timezones based on the users location. Don't use SAY TIME. Use EXEC with the SayUnixTime application and the appropriate arguments. ___ -- 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-1.2.26.tar.gz Thoughts?
I'm running 1.4 in production on the following two systems: Tyan GT20 AMD 939 dual core. openSuSE x86_64 10.1 Celeron 2.4ghz RHEL 4... cheap server from ThePlanet from what I recall they use cheap cheap cheap consumer grade stuff. Not a single crash not a single issue. I will admit we run magnitudes more traffic on 1.2 thusfar but I think we would have seen issues with 1.4 if they were. FWIW I only started testing 1.4 approx 6 months ago and put it in production approx 3 months ago. On Jan 18, 2008 3:20 PM, Ira [EMAIL PROTECTED] wrote: At 11:53 AM 1/18/2008, you wrote: Apart from the fact asterisk 1.2 is in security maintenance mode only and wont get any other bugfixes it will be ok. Please consider using 1.4 as it's the official latest stable version. Although for some of us, or at least me, no version of 1.4 has run for more than 72 hours before generating a kernel panic. I've tried about 6 versions, the early ones were good for about 10 minutes, the latest one lasted 3 days. Sadly I'm still stuck using the latest 1.2. Ira ___ -- 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] asterisk chan_sip tuning
hi, can i ask what settings do you recommend for a lot(1000-1) of different sip phones which are behind NAT(many different routers)? i have qualify=5000 nat=yes clisip show settings Reg. min duration 60 secs Reg. max duration: 3600 secs Reg. default duration: 120 secs Outbound reg. timeout: 20 secs Outbound reg. attempts: 0 asterisk 1.4 thanks --- Marek Cervenka === ___ -- 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] SAY TIME + PHPAGI + Timezone
Use the Set(TZone=blah) command in the dialplan. I.e. Set(TZone=EST5EDT) N. Nitesh Divecha wrote: Hello All, Is there any way to change the timezone on the fly? I have this little time clock program running on Asterisk system developed using PHPAGI. Currently, whenever user logs in, Asterisk will prompt the current system time using $agi-say_time(); which executes SAY TIME. Now the current timezone set on the system is PST, and I have a request to prompt multiple timezones based on the users location. First part is easy to lookup from which area code the user is calling, now the second part is to set the timezone based on the area code and prompt the users correct time. For example, if 248 (MI) user dials into the system, then time clock has to prompt EST time and if 714 (CA) user dials in then prompt PST time. Any suggestions... Thanks Cheers, Nitesh ___ -- 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] Two Asterisks behind NAT and need to link them using IAX trunk
bilal ghayyad wrote: Hi; Via OpenVPN or port forwarding is known for me, but via SSH is new for me, how I can do it and what is the difference by SSH and OpenVPN? SSH uses tcp. Openvpn, by default uses udp. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.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] OT: Call for beta testers (well... perhaps late Alpha).
Steve Totaro wrote: On Jan 18, 2008 11:00 AM, SIP [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: We've just launched the beta of a free service which is, really, still only JUST out of the alpha stages. http://www.voipmagnet.com The basic idea is this: it's an opt-in directory focused on VoIP contact info (with elements of social networking and privacy control). Again, the service is very rough, but we'd like input from the VoIP community. There are a good many things that are likely buggy, broken, or not yet implemented, but we feel that's what a beta is for. If you have any questions/concerns/issues/feedback/abuse, feel free to send it to me directly via email or post it to the list. Some things we know will be changed soon: -ability to add multiple VoIP accounts with each login (this is in the DB, but the interface elements are there yet -- we didn't like the way they looked first pass 'round) -ability to invite friends to join up so you can share contact info with friend groups without making everything (or anything) public -search optimisations (searching is functional, but a bit rough. We're looking for any and all input there, of course) We've a whole plate load of ideas for what to cram into the service, but the intent is to keep it free and provider-agnostic, but still maintain a centralised location where anyone can go and look up friends or coworkers to find out what VoIP service they're using and how to get in touch. We're open to any and all suggestions to what to add/change/fix to make this a service out of which the community will get some use. -- Neil Fusillo CEO Infinideas, inc. http://www.infinideas.com Cool idea. Kind of like LinkedIn with VoIP details. Thanks, Steve Totaro Kind of, yes. Although, we're focusing less on being yet another social network and more on being a place where you can find your VoIP friends. By this, I mean, we're not going to necessarily (at least, it's not in our plans) have an area where you can put in your favourite movies, or where you went to school, or whom you're stalking at the moment. However, there ARE plans for a possible API to mesh with contact programs. But that sort of thing is putting the cart WELL before the horse. ;) -- Neil Fusillo CEO Infinideas, inc. http://www.infinideas.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] Accessing a MySQL database and using the same db for cdr
On 1/18/08, Cyril SCETBON [EMAIL PROTECTED] wrote: Hi guys, Does someone use a mysql database for accessing data and in the same time for storing cdr ? if that is the case, which module is used ? There are two different modules for this. But it's all in asterisk-addons. For queries you would need app_addon_sql_mysql and for CDR - cdr_addon_mysql. However app_addon_sql_mysql would require you to connect/disconnect and keep track of ID's. there's also res_odbc that connects automatically, but that needs ODBC installed, and i don't like extra layers :) I'm waiting for something that would have the same interface that res_odbc but would use MySQL directly. Btw, in trunk there's improvements for Realtime engine, so you can do UPDATE's and DELETE's trough func_realtime. If you're interested - i can send you backported patch for 1.4 Regards, Atis thanks -- Cyril SCETBON ___ -- 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 -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- 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 ip phne 7911G with asterisk
On Wed, Jan 16, 2008 at 10:26:04AM -0500, Anciso, Roy wrote: Now that you have your 7911g phone up running, would you mind checking the audio quality when leaving a voicemail for on another local asterisk user from this phone? I have a 7911g and I hear loud audio taps from the phone. The 7961g phone doesn't have this issue. I'm just trying to rule out the phone. Thanks I would try if a I have more time. Nowadays I have problems with the sound quality in the conversatio. 7941g with SIP-8-0-3 firwmware sounds well but 7911G with the same firmware version sounds really bad in an Asterisk 1.2.15. I listen echo and noise taps when I talk with other sip users of the same Asterisk. I have test the phone with the same firmware againts the lastest version of Asterisk (subversion stable tree) and there is no problem with the sound quality. Do you know what is the problem in this case? Or if there is a bug in Asterisk 1.2 that could affect in that way the audio quality? Thanks, However againts the 1.4.17 version there is no problem -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Pinedo Zamalloa Sent: Wednesday, January 16, 2008 10:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cisco ip phne 7911G with asterisk On Tue, Jan 15, 2008 at 01:14:42PM +, Christian Pinedo wrote: hi, I'm trying to configure a Cisco IP Phone 7911G in order to work with Asterisk. I have loaded the 8.3.3 SIP Firmware of Cisco through a DHCP and a TFTP server. All seems ok but a file that is downloaded : term06.default.loads (I understand that is for 7906 model) instead of term11.default.loads (I understand that is for 7911 model). In any case the phone reboots well. At this moment I thought that the phone should ask the SEPmac.xml.cnf file but it asks CTLSEPmac.tlv all the time. I don't have this file in the server and it tries to download every few seconds whitout asking another file. According to what I have read this file shouldn't be neccesary and, when the phone cann't obtain it, the phone should ask SEPmac.xml.cnf. I don't know if I'm doing something bad or if it could be a issue of the firmware version. I would thank some clue. Thanks, It was a TFTP server issue. The classical TFTP server used in the unix world responds to queries with bad error codes. I finally used aTFTPD that does this well so the phone understands that there's no CTLSEP file and then asks for SEP file. -- Christian Pinedo Zamalloa (zako) PGP key at: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x828D0C80 Fingerprint: 7BFF 4105 F46B 7977 BD96 348C 1007 4FF8 828D 0C80 ___ -- 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 -- Christian Pinedo Zamalloa (zako) PGP key at: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x828D0C80 Fingerprint: 7BFF 4105 F46B 7977 BD96 348C 1007 4FF8 828D 0C80 signature.asc Description: Digital signature ___ -- 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: Call for beta testers (well... perhaps late Alpha).
On Jan 18, 2008 11:00 AM, SIP [EMAIL PROTECTED] wrote: We've just launched the beta of a free service which is, really, still only JUST out of the alpha stages. http://www.voipmagnet.com The basic idea is this: it's an opt-in directory focused on VoIP contact info (with elements of social networking and privacy control). Again, the service is very rough, but we'd like input from the VoIP community. There are a good many things that are likely buggy, broken, or not yet implemented, but we feel that's what a beta is for. If you have any questions/concerns/issues/feedback/abuse, feel free to send it to me directly via email or post it to the list. Some things we know will be changed soon: -ability to add multiple VoIP accounts with each login (this is in the DB, but the interface elements are there yet -- we didn't like the way they looked first pass 'round) -ability to invite friends to join up so you can share contact info with friend groups without making everything (or anything) public -search optimisations (searching is functional, but a bit rough. We're looking for any and all input there, of course) We've a whole plate load of ideas for what to cram into the service, but the intent is to keep it free and provider-agnostic, but still maintain a centralised location where anyone can go and look up friends or coworkers to find out what VoIP service they're using and how to get in touch. We're open to any and all suggestions to what to add/change/fix to make this a service out of which the community will get some use. -- Neil Fusillo CEO Infinideas, inc. http://www.infinideas.com Cool idea. Kind of like LinkedIn with VoIP details. Thanks, Steve Totaro ___ -- 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] Zaptel timing on TE405P
On Fri, Jan 18, 2008 at 05:44:12PM +0200, Atis Lezdins wrote: On 1/17/08, Atis Lezdins [EMAIL PROTECTED] wrote: On 1/17/08, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Jan 17, 2008 at 03:09:59PM +0200, Atis Lezdins wrote: Hi, I'm wondering why zttest shows Best: 99.976 -- Worst: 99.967 -- Average: 99.971469, Difference: 99.971469 Shouldn't it be 100% as timing is hardware and comes from PRI? Am I missing some kernel config? It may be slightly different. Your system clock may be slightly off. But more importantly, zttest doesn't start and stop messuring time at exactly the right spot. Anything i can improve? I think - zttest should do it correctly, as manpage says - definite pass is 100% or 99.99% I'm just having some issues with faxing, so i thought this could be a problem. Ping. Any ideas what i could do to improve timing accuracy? Some kernel options? Newer kernel? Currently I have kernel from RPM: The question is: How to improve the meassurment of timing. Also, some report that Steve Underwood's sliptest is a useful tool for that. If you find it useful, I have a small patch that makes it slightly more usable. http://www.soft-switch.org/downloads/sliptest.c -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] Accessing a MySQL database and using the same db for cdr
Ciao Cyril, Does someone use a mysql database for accessing data and in the same time for storing cdr ? if that is the case, which module is used ? thanks Which kind of data are you talking about? I suppose that you mean that you want to store non-Asterisk related data and CDR data in the same database. In this case, to handle CDR data you should use the cdr_mysql module, from asterisk-addons (http://www.voip-info.org/wiki-Asterisk+cdr+mysql), storing CDR data in a table of your choice. Then you should use the DB for your data exactly as you would do if Asterisk weren't storing his data. HTH, -- Dr. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l. Web: www.x-voice.it - Tel: +39 (0) 95 7224945 ___ -- 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] Two Asterisks behind NAT and need to link them using IAX trunk
Am Freitag, den 18.01.2008, 04:21 -0800 schrieb bilal ghayyad: Hi; Via OpenVPN or port forwarding is known for me, but via SSH is new for me, how I can do it and what is the difference by SSH and OpenVPN? In principle both use a packet stream (SSH is TCP, OpenVPN is TCP or UDP) for encapsulating IP packets. The main difference is that SSH port forwarding forwards the packet data, but not the header: The packet is stripped at side A and a seemingly different TCP connection is established on side B. This also implies the main limitation of SSH, that it is restricted to tunneling TCP (afaik). OpenVPN in contrast takes entire IP packets, applies routing and tunnels the entire packet through. You can tunnel any IP traffic through OpenVPN, and the remote side IP address will persist. (You can even tunnel IPX or Appletalk, if using the BRIDGE mode with virtual TAP interfaces). Basically OpenVPN appears to the tunnel endpoint as a virtual wire that behaves like an ethernet port. OpenVPN is far more flexible when it comes to network restrictions. On the other hand the SSH main idea is not VPN but secure shell access :) For VoIP I'd imagine SSH is quite impractical, if usable at all. Most likely the TCP-only restriction will make life difficult. SIP over OpenVPN works - I used it to tunnel from a trip to California to my Asterisk back home in Germany. The voice quality was a bit poor, but this might also relate to the WLAN and the multi-hop-internet route in between. Speaking generally, of course an aditional layer (which both OpenVPN and SSH introduce) does not improve the signal path quality, or latency, or everything. I have read recommendations to use OpenVPN in UDP mode to reduce packetizing problems which would result in choppy sound as well. No comparison numbers available here though. BR Anselm ___ -- 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] SAY TIME + PHPAGI + Timezone
Hello All, Is there any way to change the timezone on the fly? I have this little time clock program running on Asterisk system developed using PHPAGI. Currently, whenever user logs in, Asterisk will prompt the current system time using $agi-say_time(); which executes SAY TIME. Now the current timezone set on the system is PST, and I have a request to prompt multiple timezones based on the users location. First part is easy to lookup from which area code the user is calling, now the second part is to set the timezone based on the area code and prompt the users correct time. For example, if 248 (MI) user dials into the system, then time clock has to prompt EST time and if 714 (CA) user dials in then prompt PST time. Any suggestions... Thanks Cheers, Nitesh ___ -- 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-1.2.26.tar.gz Thoughts?
At 11:53 AM 1/18/2008, you wrote: Apart from the fact asterisk 1.2 is in security maintenance mode only and wont get any other bugfixes it will be ok. Please consider using 1.4 as it's the official latest stable version. Although for some of us, or at least me, no version of 1.4 has run for more than 72 hours before generating a kernel panic. I've tried about 6 versions, the early ones were good for about 10 minutes, the latest one lasted 3 days. Sadly I'm still stuck using the latest 1.2. Ira What type of Asterisk setup do you have? While my setup is not a large commercial setup I have seen asterisk 1.4 with a few calls going through it at once last for weeks if not months before it was restarted. Just curious. ___ -- 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-1.2.26.tar.gz Thoughts?
On 14:34, Fri 18 Jan 08, Ryan Burke wrote: At 11:53 AM 1/18/2008, you wrote: Apart from the fact asterisk 1.2 is in security maintenance mode only and wont get any other bugfixes it will be ok. Please consider using 1.4 as it's the official latest stable version. Although for some of us, or at least me, no version of 1.4 has run for more than 72 hours before generating a kernel panic. I've tried about 6 versions, the early ones were good for about 10 minutes, the latest one lasted 3 days. Sadly I'm still stuck using the latest 1.2. Ira What type of Asterisk setup do you have? While my setup is not a large commercial setup I have seen asterisk 1.4 with a few calls going through it at once last for weeks if not months before it was restarted. Just curious. I follow asterisk 1.4 svn. I have around 25 customers with avg 10 phones and roughly 20 extensions with avg 10 priorities in every exten. This is a pure voip setup with IAX2 connections to 4 different ITSP's and SIP to the phones. There have been some issues in the early versions, but it's fine now. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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] New Polycom Provisioning Tool Released with BugFix
Polycom Provisioning Tool Updated. I made a bug fix that was reported, which was causing the directory creator to not work when there was an invalid character in the filename of the csv. I have also posted an FAQ: http://www.wintrisk.com/ppt.html#FAQ Download the new one, and tell me what you think! It's free, and mildly useful! http://www.wintrisk.com/ppt.html Yours, Michael Munger, dCAP 404-438-2128 [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] asterisk-1.2.26.tar.gz Thoughts?
On Fri, Jan 18, 2008 at 12:20:56PM -0800, Ira wrote: At 11:53 AM 1/18/2008, you wrote: Apart from the fact asterisk 1.2 is in security maintenance mode only and wont get any other bugfixes it will be ok. Please consider using 1.4 as it's the official latest stable version. Although for some of us, or at least me, no version of 1.4 has run for more than 72 hours before generating a kernel panic. I've tried about 6 versions, the early ones were good for about 10 minutes, the latest one lasted 3 days. Sadly I'm still stuck using the latest 1.2. Kernel panics can be caused by buggy kernel code and / or bad hardware. Buggy userspace should not (by definition) be able to cause them. If userspace can, it's a kernel bug. So can you be more specific about those panics? Do you have traces from them? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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-1.2.26.tar.gz Thoughts?
I would suspect that your hardware is the cause of your problems. Running a production PBX system on a discarded desktop system is a /really/ bad idea. I would seriously consider an upgrade to your hardware. Ira wrote: At 12:34 PM 1/18/2008, you wrote: Although for some of us, or at least me, no version of 1.4 has run for more than 72 hours before generating a kernel panic. I've tried about 6 versions, the early ones were good for about 10 minutes, the latest one lasted 3 days. Sadly I'm still stuck using the latest 1.2. Ira What type of Asterisk setup do you have? While my setup is not a large commercial setup I have seen asterisk 1.4 with a few calls going through it at once last for weeks if not months before it was restarted. Just curious. 1ghz Celeron, 1 gig ram, 120gb HD. An HP home desktop discarded by a client 2 year old Digium 4 FXO port card using only 3 ports and the Digium HP echo can 3 analog lines in 2 SIP lines in most outgoing via SIP Most incoming via analog phones are all Aastra 480i-CT Dial plan is hand written, likely a bit convoluted, but it's hard to avoid that. Seems like the panics were mostly to do with ZAP The internet runs over 192.168.0.XXX the phones run on 192.168.233.XXX The two networks are completely separate until they reach the router connected to the world. The only problem I have with the most current 1.2 is every month or three it thinks one of the phones has 5 active lines going and stops sending calls to it, restart gracefully and all is well again. Ira ___ -- 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
