Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
I wonder if there are any major obstacles for upgrading. Just tried an in-place upgrade on my home box : make[1]: Leaving directory `/usr/src/asterisk-addons-1.4.5' for x in app_addon_sql_mysql.so app_saycountpl.so cdr_addon_mysql.so res_config_mysql.so; do /usr/bin/install -c -m 755 $x /usr/lib/asterisk/modules ; done /usr/bin/install: cannot stat `app_addon_sql_mysql.so': No such file or directory /usr/bin/install: cannot stat `cdr_addon_mysql.so': No such file or directory /usr/bin/install: cannot stat `res_config_mysql.so': No such file or directory make: *** [install] Error 1 And the asterisk console is flooded with these errors : [Dec 21 10:10:58] WARNING[22897]: chan_sip.c:6707 determine_firstline_parts: Bad request protocol Packet [Dec 21 10:10:58] WARNING[22897]: chan_sip.c:6707 determine_firstline_parts: Bad request protocol Packet So for the next time to come i'll turn back to 1.2 :) ___ --Bandwidth and Colocation Provided 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] txfax not working with spandsp
the attached log with verbose=6 and debug=6 refers. we've got a sangoma A104 (no hwec) with PRI ports 1 3 loopbacked to each other. we're trying to have txfax send out on one of those pri ports with rxfax listening on the other side, hence having asterisk send a fax to itself. we however have bad, and i mean really bad (10%) success rates. we're currently using asterisk 1.2.24 with spandsp 0.0.4-20071214 (snapshot of 14/12/07) and we keep getting Fax send not successful - result (25) No response after sending a page. errors. ECM is turned on in both app_txfax.c and app_rxfax.c. from what we gather just reading the code, time T4 expires in txfax because apparently rxfax is not sending a response back out, and thus after the maximum message retries (3) txfax just drops the call, leading rxfax to say that the call was dropped prematurely. does anyone know what's going on here, and if there is a version of spandsp which could work in this scenario ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ Dec 21 18:32:05 VERBOSE[205] logger.c: -- Attempting call on Zap/g1/1002 for [EMAIL PROTECTED]:1 (Retry 1) Dec 21 18:32:05 DEBUG[205] chan_zap.c: Using channel 1 Dec 21 18:32:05 VERBOSE[205] logger.c: -- Requested transfer capability: 0x00 - SPEECH Dec 21 18:32:05 DEBUG[205] devicestate.c: Changing state for Zap/1 - state 2 (In use) Dec 21 18:32:05 DEBUG[205] devicestate.c: Changing state for Zap/1 - state 2 (In use) Dec 21 18:32:05 DEBUG[205] app_queue.c: Device 'Zap/1' changed to state '2' (In use) but we don't care because they're not a member of any queue. Dec 21 18:32:05 DEBUG[205] app_queue.c: Device 'Zap/1' changed to state '2' (In use) but we don't care because they're not a member of any queue. Dec 21 18:32:05 VERBOSE[205] logger.c: -- Accepting call from '' to '1002' on channel 0/1, span 2 Dec 21 18:32:05 DEBUG[205] chan_zap.c: No echo cancellation requested Dec 21 18:32:05 DEBUG[205] devicestate.c: Changing state for Zap/32 - state 2 (In use) Dec 21 18:32:05 DEBUG[205] pbx.c: Launching 'Answer' Dec 21 18:32:05 VERBOSE[205] logger.c: -- Executing Answer(Zap/32-1, ) in new stack Dec 21 18:32:05 DEBUG[205] pbx.c: Launching 'Wait' Dec 21 18:32:05 VERBOSE[205] logger.c: -- Executing Wait(Zap/32-1, ) in new stack Dec 21 18:32:05 DEBUG[205] pbx.c: Launching 'Set' Dec 21 18:32:05 VERBOSE[205] logger.c: -- Executing Set(Zap/32-1, FAXFILE=/tmp/FAX-1198233125.1.tiff) in new stack Dec 21 18:32:05 DEBUG[205] pbx.c: Function result is '1002' Dec 21 18:32:05 DEBUG[205] pbx.c: Launching 'Set' Dec 21 18:32:05 VERBOSE[205] logger.c: -- Executing Set(Zap/32-1, NEWFAXFILE=/var/spool/asterisk/fax/FAX-1002--11982331251198233125.1.tiff) in new stack Dec 21 18:32:05 DEBUG[205] pbx.c: Launching 'RxFAX' Dec 21 18:32:05 VERBOSE[205] logger.c: -- Executing RxFAX(Zap/32-1, /tmp/FAX-1198233125.1.tiff|debug) in new stack Dec 21 18:32:05 DEBUG[205] app_queue.c: Device 'Zap/32' changed to state '2' (In use) but we don't care because they're not a member of any queue. Dec 21 18:32:05 DEBUG[205] devicestate.c: Changing state for Zap/32 - state 2 (In use) Dec 21 18:32:05 DEBUG[205] channel.c: Set channel Zap/32-1 to read format slin Dec 21 18:32:05 DEBUG[205] channel.c: Set channel Zap/32-1 to write format slin Dec 21 18:32:05 DEBUG[205] app_queue.c: Device 'Zap/32' changed to state '2' (In use) but we don't care because they're not a member of any queue. Dec 21 18:32:05 DEBUG[205] chan_zap.c: Queuing frame from PRI_EVENT_PROCEEDING on channel 0/1 span 1 Dec 21 18:32:05 DEBUG[205] chan_zap.c: No echo cancellation requested Dec 21 18:32:05 VERBOSE[205] logger.c: Channel Zap/1-1 was answered. Dec 21 18:32:05 DEBUG[205] pbx.c: Launching 'Answer' Dec 21 18:32:05 VERBOSE[205] logger.c: -- Executing Answer(Zap/1-1, ) in new stack Dec 21 18:32:05 DEBUG[205] pbx.c: Launching 'Set' Dec 21 18:32:05 VERBOSE[205] logger.c: -- Executing Set(Zap/1-1, CDR(userfield)=FAX-1) in new stack Dec 21 18:32:05 DEBUG[205] pbx.c: Launching 'TxFAX' Dec 21 18:32:05 VERBOSE[205] logger.c: -- Executing TxFAX(Zap/1-1, /var/spool/asterisk/outgoing_fax/page.1.1.tiff|caller|debug) in new stack Dec 21 18:32:05 DEBUG[205] channel.c: Set channel Zap/1-1 to read format slin Dec 21 18:32:05 DEBUG[205] channel.c: Set channel Zap/1-1 to write format slin Dec 21 18:32:05 DEBUG[205] devicestate.c: Changing state for Zap/1 - state 2 (In use) Dec 21 18:32:05
Re: [asterisk-users] ip phone suggestion for Asia?
Check out yntx www.yntx.com fear prices and recides in Asia and iss it sip on asteriks they will do ! try to buy one to trye it out before buying fore hole company.. /MVH Fille ___ --Bandwidth and Colocation Provided 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] Resposta automàtica (was: a sterisk-users Digest, Vol 41, Issue 67)
Salutacions!!! Si has arribat fins aqu, s perqu he configurat el meu correu perqu et retorni un resposta automtica ja que jo estar fora durent un parell de mesos Vaig de vacances a Sibria... Jordi. ___ --Bandwidth and Colocation Provided 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] Snom 370 buton Recordings
Hello all, I am using the Snom 370 phone with firmware Snom370-SIP 7.0.17* *connected to an asterisk 1.2.14 and I can't record any calls using the Recording button on this phone. The extension I configured on this phone has the values Recording on demand, an the voicemail enabled. I am using FreePBX to manage my PBX. How should I configure the Function keys to make this work? Anybody have made this button works on this phone? How? Any clue will be welcomed. 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] ip phone suggestion for Asia?
On Dec 21, 2007 12:37 AM, d tbsky [EMAIL PROTECTED] wrote: hi gnubie: snom seems has some re-brand ip phones. do they use the same firmware? if they are the same, i don't understand why snom do this.. If I'm not mistaken, they use the same firmware. I don't know about Aztech and SNOM business relationship, if there is. But for sure, it is still a SNOM phone. =) GNUbie ___ --Bandwidth and Colocation Provided 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] txfax not working with spandsp
On Fri, 2007-12-21 at 18:41 +0800, Dinesh Nair wrote: the attached log with verbose=6 and debug=6 refers. we've got a sangoma A104 (no hwec) with PRI ports 1 3 loopbacked to each other. we're trying to have txfax send out on one of those pri ports with rxfax listening on the other side, hence having asterisk send a fax to itself. we however have bad, and i mean really bad (10%) success rates. we're currently using asterisk 1.2.24 with spandsp 0.0.4-20071214 (snapshot of 14/12/07) and we keep getting Fax send not successful - result (25) No response after sending a page. errors. ECM is turned on in both app_txfax.c and app_rxfax.c. from what we gather just reading the code, time T4 expires in txfax because apparently rxfax is not sending a response back out, and thus after the maximum message retries (3) txfax just drops the call, leading rxfax to say that the call was dropped prematurely. does anyone know what's going on here, and if there is a version of spandsp which could work in this scenario ? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you have any timing issues such as slips or bi-polar violations taking place. It sounds like there are dropped packets or something. dave ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming CID change
Quick and dirty too hungover to think/search... ;) Sorry list AsteriskBoxA - SBC/PSTN voodoo - world wide Interweb - AsteriskBoxB Account on BoxA UsernameJohn12125551212 Account on BoxB UsernameJohn102 They're both the same users, had to do some funky trunking (managed firewall provider is playing not in my backyard games)... So anyway, I need to take specific calls from BoxA with the CID of UsernameJohn12125551212 and switch it to UsernameJohn102 on BoxB Make sense? The incoming CID will ALWAYS be 12125551212 ... J. Oquendo SGFA #579 (FW+VPN v4.1) SGFE #574 (FW+VPN v4.1) I hear much of people's calling out to punish the guilty, but very few are concerned to clear the innocent. Daniel Defoe http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0xF684C42E smime.p7s Description: S/MIME Cryptographic 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] ip phone suggestion for Asia?
hi Fredrik : thanks for your information. after checking yntx manuals, i found i have one phone in my hand, which has the same firmware with yntx phones. although it is a different brand and looks different. the phone's basic function is ok, but we need some advanced functions like xml phonebook. i hope these china phones would catch up quickly so we all can have better, cheaper phones. Regards, tbskyd 2007/12/21, Fredrik Söderlund [EMAIL PROTECTED]: Check out yntx www.yntx.com fear prices and recides in Asia and iss it sip on asteriks they will do ! try to buy one to trye it out before buying fore hole company.. /MVH Fille ___ --Bandwidth and Colocation Provided 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] Upgrade to Asterisk 1.4 - it's one year's old!
remco, I just had the same problem/error on my CLI when I added a polycom shoretel IP-100 phone to my network and enabled mgcp... couldn't figure out how to get that working yet... I don't think it is related to 1.4 as I have been running 1.4 has been running for over a year now without that error... I would look somewhere else... daveC [Dec 21 08:51:32] WARNING[16742]: chan_sip.c:6620 determine_firstline_parts: Bad request protocol [EMAIL PROTECTED]] MGCP 1.0 Remco Barendse wrote: I wonder if there are any major obstacles for upgrading. Just tried an in-place upgrade on my home box : make[1]: Leaving directory `/usr/src/asterisk-addons-1.4.5' for x in app_addon_sql_mysql.so app_saycountpl.so cdr_addon_mysql.so res_config_mysql.so; do /usr/bin/install -c -m 755 $x /usr/lib/asterisk/modules ; done /usr/bin/install: cannot stat `app_addon_sql_mysql.so': No such file or directory /usr/bin/install: cannot stat `cdr_addon_mysql.so': No such file or directory /usr/bin/install: cannot stat `res_config_mysql.so': No such file or directory make: *** [install] Error 1 And the asterisk console is flooded with these errors : [Dec 21 10:10:58] WARNING[22897]: chan_sip.c:6707 determine_firstline_parts: Bad request protocol Packet [Dec 21 10:10:58] WARNING[22897]: chan_sip.c:6707 determine_firstline_parts: Bad request protocol Packet So for the next time to come i'll turn back to 1.2 :) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --Bandwidth and Colocation Provided 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] ip phone suggestion for Asia?
hi mkezys: ok. i will add linksys to our testing list. but cisco tend to lock things. can we get firmware for linksys easily ? or we must pay like cisco routers and switches? 2007/12/21, Mindaugas Kezys [EMAIL PROTECTED]: Do not forget to evaluate Linksys SPA phones. Best I tried and not expensive. We use them (SPA942) in our company. Everybody's happy. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of d tbsky Sent: Thursday, December 20, 2007 6:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ip phone suggestion for Asia? Hi: i am surveying ip phones for our company. we will use them with asterisk. we have office in taiwan, hong kong,singapore and china. cisco and polycom are too expensive for us. we try several china brand ip phones. they are all cheap and some of them have good quality. but most of them won't offer future firmware support, which we think it's important for ip phones. searching in the mail list, we found aastra is good, but they don't sale to asia. grandstream looks good also.there are many grandstream users in the list, can someone share any good or bad experience about grandstream today? if there are other good choice, please tell us!! thanks a lot for your help!! Regards, tbskyd ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 7970 CTLFile.tlv?
I believe you can create a blank file to keep the phone from complaining. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Rubenstein Sent: Friday, December 21, 2007 10:16 AM To: Asterisk -Users Subject: [asterisk-users] 7970 CTLFile.tlv? I've got a Cisco 7970 that's not completing its network registration to Asterisk. The Registering message stays on the screen (with the moving time wheel). After a few minutes, the onscreen message flashes Updating CTL then Loading..., then the status messages update with: No valid CAPF server File Not Found: CTLFile.tlv No CTL installed SEPMAC.cnf.xml (where MAC is the phone's MAC addr minus :s) before repeating the cycle (forever). Where can I get a CTLFile.tlv , or remove the requirement for it? Or is there another way to fix this problem? TIA. Asterisk v1.2.9.1-BRIstuffed-0.3.0-PRE-1q SCCP firmware Load File: TERM70.7-0-1-0s App Load ID: Jar70.2-9-0-117.sbn JVM Load ID: CVM70.2-0-0-112.sbn OS Load ID: cnu70.2-7-4-134.sbn Boot Load ID: 7970_64060118.bin -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided 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] 7970 CTLFile.tlv?
I've got a Cisco 7970 that's not completing its network registration to Asterisk. The Registering message stays on the screen (with the moving time wheel). After a few minutes, the onscreen message flashes Updating CTL then Loading..., then the status messages update with: No valid CAPF server File Not Found: CTLFile.tlv No CTL installed SEPMAC.cnf.xml (where MAC is the phone's MAC addr minus :s) before repeating the cycle (forever). Where can I get a CTLFile.tlv , or remove the requirement for it? Or is there another way to fix this problem? TIA. Asterisk v1.2.9.1-BRIstuffed-0.3.0-PRE-1q SCCP firmware Load File: TERM70.7-0-1-0s App Load ID: Jar70.2-9-0-117.sbn JVM Load ID: CVM70.2-0-0-112.sbn OS Load ID: cnu70.2-7-4-134.sbn Boot Load ID: 7970_64060118.bin -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided 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 SIP handling - why 491 Request Pending response
Hi, I have the following situation I use asterisk as o gateway between networks. What is the reason for such response? What are the criteria for such evaluation? SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.129.74:5160;branch=z9hG4bK17c3.17db29e7.0;received= 192.168.129.74 Via: SIP/2.0/UDP 192.168.129.74;branch=z9hG4bK17c3.23083974.0 Via: SIP/2.0/UDP 192.168.129.74:5070;branch=z9hG4bK5b33ae78;rport=5070 From: IPFon sip:[EMAIL PROTECTED]:5070;tag=as7217acbc To: sip:[EMAIL PROTECTED];tag=as7217acbc Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 CHeers tomasz ___ --Bandwidth and Colocation Provided 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] Send SIP 100 Trying instead of 183 Session Progress
Hi, I have a Asterisk that connects to the PSTN via a PRI. After Asterisk sends the setup message it immediately sends a 183 Session Progress. Is there a way I can change it so that it sends a 100 Trying instead? Because I am having some issues with a equipment where it does not play a busy tone as a result of sending a 183 Session Progress then the 486 Busy. Thanks Remi ___ --Bandwidth and Colocation Provided 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] 7970 CTLFile.tlv?
You don't need the .tlv file. It's optional, and will be skipped if it cannot be found. Your problem is elsewhere. I've found that the 7970s are very finicky. I've never had luck with the SEPMAC.cnf.xml - only XmlDefault.cnf.xml (case may vary - check your tftp logs) Matthew Rubenstein wrote: I've got a Cisco 7970 that's not completing its network registration to Asterisk. The Registering message stays on the screen (with the moving time wheel). After a few minutes, the onscreen message flashes Updating CTL then Loading..., then the status messages update with: No valid CAPF server File Not Found: CTLFile.tlv No CTL installed SEPMAC.cnf.xml (where MAC is the phone's MAC addr minus :s) before repeating the cycle (forever). Where can I get a CTLFile.tlv , or remove the requirement for it? Or is there another way to fix this problem? TIA. Asterisk v1.2.9.1-BRIstuffed-0.3.0-PRE-1q SCCP firmware Load File: TERM70.7-0-1-0s App Load ID: Jar70.2-9-0-117.sbn JVM Load ID: CVM70.2-0-0-112.sbn OS Load ID: cnu70.2-7-4-134.sbn Boot Load ID: 7970_64060118.bin -- Jason Parker Digium ___ --Bandwidth and Colocation Provided 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 7961 new firmware stops readingconfiguration files
I contacted one of the list users and they sent me their configuration files. I used it as a template and it worked with my phone, so I'll be sure to put it back up on the Wiki. Chad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Preston Edwards Sent: December 20, 2007 6:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7961 new firmware stops readingconfiguration files Chad, I had the same problem when upgrading to some of the newer firmware. The newer firmware gets even pickier (if that's even possible) about the config files. Go the phone's webpage and look at the debug log. It will show you where it's not parsing correctly. I'm not in front of my phone now so I can't look, but I remember it getting upset about networkLocale or userLocale or something of that nature, so I just removed that section of the XML code and it loaded fine. Good luck, Preston Edwards __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email _ --Bandwidth and Colocation Provided 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] ip phone suggestion for Asia?
Hi How about trying the Snom phones? They are good and feature rich with regular firmware upgrades. Thanks Neel d tbsky [EMAIL PROTECTED] wrote: hi Fredrik : thanks for your information. after checking yntx manuals, i found i have one phone in my hand, which has the same firmware with yntx phones. although it is a different brand and looks different. the phone's basic function is ok, but we need some advanced functions like xml phonebook. i hope these china phones would catch up quickly so we all can have better, cheaper phones. Regards, tbskyd 2007/12/21, Fredrik Söderlund : Check out yntx www.yntx.com fear prices and recides in Asia and iss it sip on asteriks they will do ! try to buy one to trye it out before buying fore hole company.. /MVH Fille ___ --Bandwidth and Colocation Provided 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 - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ --Bandwidth and Colocation Provided 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] Telco MWI Detection on TDM400 Interface?
Jim Duda wrote: Thanks Russell, that's what I'm looking for. You're welcome! Any idea when this will become part an official asterisk release? It will be a part of Asterisk 1.6. (and BE C.1 ...) -- Russell Bryant Senior Software Engineer Open Source Team Lead 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
Re: [asterisk-users] Asterisk SIP handling - why 491 Request Pending response
What is the reason for such response? SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.129.74 :5160;branch=z9hG4bK17c3.17db29e7.0;received=192.168.129.74 Via: SIP/2.0/UDP 192.168.129.74 ;branch=z9hG4bK17c3.23083974.0 Via: SIP/2.0/UDP 192.168.129.74:5070;branch=z9hG4bK5b33ae78;rport=5070 From: IPFon sip:[EMAIL PROTECTED]:5070;tag=as7217acbc To: sip:[EMAIL PROTECTED];tag=as7217acbc Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Asterisk will send a 491 Request Pending when it is currently processing an INIVTE on a particular call and it gets another INVITE that isn't a retransmission. ___ --Bandwidth and Colocation Provided 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] Polycom 330 beep on new VM
Hi, I have a Polycom 330 that emits a beep every 30s or so when there is a message waiting. Is there a way to disable that? It is pretty annoying. Regards, Ugo ___ --Bandwidth and Colocation Provided 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] Realtime: Should I say or should I go (now) ?
I'm working on a 500 seats Asterisk project. I'm wondering whether or not I should consider using Asterisk Realtime and a database to manage phones registrations. As far as stability goes, I've had no problem with realtime. In fact I've run a nationwide VoIP provider with asterisk using realtime. The main consideration I would make for using/not using it is how dynamic the system is. If you have 500 seats that aren't likely to change and aren't designing an interface to actually manage those users, then go static. Less complexity, less chance for failure (you don't have to worry about db servers, etc.). On the other hand, if it is for a situation where you are constantly adding phones, changing caller-ids, swapping phones and their extensions, etc. Realtime is the only way to go. And a free hint: if you are going to have to do anything that resembles number porting, swapping extensions, etc.--don't use extensions/phone numbers as SIP usernames. You have to regenerate config files, etc. Make your SIP usernames meaningless and use func_odbc to look up what extension is tied to which device. Terry ___ --Bandwidth and Colocation Provided 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] ip phone suggestion for Asia?
Hi! d tbsky wrote: ok. i will add linksys to our testing list. but cisco tend to lock things. can we get firmware for linksys easily ? or we must pay like cisco routers and switches? You can download latest firmware from linksys.com, also here is firmware release notes with full changes list. There is some support issues: support of VoIP devices only for itsp, but community can give answer on very-very advanced questions. -- Best regards, Igor A. Goncharovsky ICQ: 648337 mailto: [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] 7970 CTLFile.tlv?
I did (/tftpboot/CTLFile.tlv), but the phone keeps complaining (no change). On Fri, 2007-12-21 at 10:25 -0500, Anciso, Roy wrote: I believe you can create a blank file to keep the phone from complaining. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Rubenstein Sent: Friday, December 21, 2007 10:16 AM To: Asterisk -Users Subject: [asterisk-users] 7970 CTLFile.tlv? I've got a Cisco 7970 that's not completing its network registration to Asterisk. The Registering message stays on the screen (with the moving time wheel). After a few minutes, the onscreen message flashes Updating CTL then Loading..., then the status messages update with: No valid CAPF server File Not Found: CTLFile.tlv No CTL installed SEPMAC.cnf.xml (where MAC is the phone's MAC addr minus :s) before repeating the cycle (forever). Where can I get a CTLFile.tlv , or remove the requirement for it? Or is there another way to fix this problem? TIA. Asterisk v1.2.9.1-BRIstuffed-0.3.0-PRE-1q SCCP firmware Load File: TERM70.7-0-1-0s App Load ID: Jar70.2-9-0-117.sbn JVM Load ID: CVM70.2-0-0-112.sbn OS Load ID: cnu70.2-7-4-134.sbn Boot Load ID: 7970_64060118.bin -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided 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] Upgrade to Asterisk 1.4 - it's one year's old!
It may be a year old.. but until Digium is drinking their own dog food.. I won't be using it. On Dec 21, 2007 9:26 AM, dave cantera [EMAIL PROTECTED] wrote: remco, I just had the same problem/error on my CLI when I added a polycom shoretel IP-100 phone to my network and enabled mgcp... couldn't figure out how to get that working yet... I don't think it is related to 1.4 as I have been running 1.4 has been running for over a year now without that error... I would look somewhere else... daveC [Dec 21 08:51:32] WARNING[16742]: chan_sip.c:6620 determine_firstline_parts: Bad request protocol [EMAIL PROTECTED] [EMAIL PROTECTED]] MGCP 1.0 Remco Barendse wrote: I wonder if there are any major obstacles for upgrading. Just tried an in-place upgrade on my home box : make[1]: Leaving directory `/usr/src/asterisk-addons-1.4.5' for x in app_addon_sql_mysql.so app_saycountpl.so cdr_addon_mysql.so res_config_mysql.so; do /usr/bin/install -c -m 755 $x /usr/lib/asterisk/modules ; done /usr/bin/install: cannot stat `app_addon_sql_mysql.so': No such file or directory /usr/bin/install: cannot stat `cdr_addon_mysql.so': No such file or directory /usr/bin/install: cannot stat `res_config_mysql.so': No such file or directory make: *** [install] Error 1 And the asterisk console is flooded with these errors : [Dec 21 10:10:58] WARNING[22897]: chan_sip.c:6707 determine_firstline_parts: Bad request protocol Packet [Dec 21 10:10:58] WARNING[22897]: chan_sip.c:6707 determine_firstline_parts: Bad request protocol Packet So for the next time to come i'll turn back to 1.2 :) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com856.380.0894 ___ --Bandwidth and Colocation Provided 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] 7970 CTLFile.tlv?
Try to change your verbose setting of tftboot server and look what file is asked for exactly Matthew Rubenstein schreef: I did (/tftpboot/CTLFile.tlv), but the phone keeps complaining (no change). On Fri, 2007-12-21 at 10:25 -0500, Anciso, Roy wrote: I believe you can create a blank file to keep the phone from complaining. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Rubenstein Sent: Friday, December 21, 2007 10:16 AM To: Asterisk -Users Subject: [asterisk-users] 7970 CTLFile.tlv? I've got a Cisco 7970 that's not completing its network registration to Asterisk. The Registering message stays on the screen (with the moving time wheel). After a few minutes, the onscreen message flashes Updating CTL then Loading..., then the status messages update with: No valid CAPF server File Not Found: CTLFile.tlv No CTL installed SEPMAC.cnf.xml (where MAC is the phone's MAC addr minus :s) before repeating the cycle (forever). Where can I get a CTLFile.tlv , or remove the requirement for it? Or is there another way to fix this problem? TIA. Asterisk v1.2.9.1-BRIstuffed-0.3.0-PRE-1q SCCP firmware Load File: TERM70.7-0-1-0s App Load ID: Jar70.2-9-0-117.sbn JVM Load ID: CVM70.2-0-0-112.sbn OS Load ID: cnu70.2-7-4-134.sbn Boot Load ID: 7970_64060118.bin ___ --Bandwidth and Colocation Provided 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] 7970 CTLFile.tlv?
I've got in.atftpd running out of inetd: - /etc/inetd.conf tftpdgram udp waitnobody /usr/sbin/tcpd /usr/sbin/in.tftpd --logfile /tmp/atftpd.log --pidfile /tmp/atftpd.pid --tftpd-timeout 300 --retry-timeout 5 --mcast-port 1758 --mcast-addr 239.255.0.0-255 --maxthread 100 --verbose=5 --no-blksize /tftpboot --- But even when I run use a tftp client from a host on the inside network to retrieve the SEPMAC.cnf.xml file successfully, the /tmp/atftpd.log file is never touched, nor is the /tmp/atftpd.pid ever created. Even if I (touch /tmp/atftpd.log; chown nobody.nogroup /tmp/atftpd.log) the status files are untouched. But I am getting the requested file. Also, what do I do to use an XmlDefault.cnf.xml file? Just rename the SEP.cnf.xml file to that? I also saw on the Web someone who had my problem with the 7970, but cryptically noted that they solved their problem which was wrong platform newline terminations. What chars does the 7970 need for its conf files newlines to be? On Fri, 2007-12-21 at 09:47 -0600, Jason Parker wrote: You don't need the .tlv file. It's optional, and will be skipped if it cannot be found. Your problem is elsewhere. I've found that the 7970s are very finicky. I've never had luck with the SEPMAC.cnf.xml - only XmlDefault.cnf.xml (case may vary - check your tftp logs) Matthew Rubenstein wrote: I've got a Cisco 7970 that's not completing its network registration to Asterisk. The Registering message stays on the screen (with the moving time wheel). After a few minutes, the onscreen message flashes Updating CTL then Loading..., then the status messages update with: No valid CAPF server File Not Found: CTLFile.tlv No CTL installed SEPMAC.cnf.xml (where MAC is the phone's MAC addr minus :s) before repeating the cycle (forever). Where can I get a CTLFile.tlv , or remove the requirement for it? Or is there another way to fix this problem? TIA. Asterisk v1.2.9.1-BRIstuffed-0.3.0-PRE-1q SCCP firmware Load File: TERM70.7-0-1-0s App Load ID: Jar70.2-9-0-117.sbn JVM Load ID: CVM70.2-0-0-112.sbn OS Load ID: cnu70.2-7-4-134.sbn Boot Load ID: 7970_64060118.bin -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided 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 330 beep on new VM
This is pretty easy to suppress using the configuration files. Check: http://www.voip-info.org/wiki/view/Polycom+SoundPoint+IP+MWI+audio On Dec 21, 2007 11:55 AM, Ugo Bellavance [EMAIL PROTECTED] wrote: Hi, I have a Polycom 330 that emits a beep every 30s or so when there is a message waiting. Is there a way to disable that? It is pretty annoying. Regards, Ugo ___ --Bandwidth and Colocation Provided 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] best way for night ringer??
Asterisk 1.2.13 I am trying to figure out the best way for a night bell at work. Note: I have no spare buttons available on the phones. But I do have two lines and two park positions as buttons. Option 1 (easiest and the one I just implemented) When asterisk is in night mode, Connect to IVR, List all options and then if they dial 0 or timeout, ring every phone in the building. Option 2 (This would require user training) When asterisk is in night mode, List all options and then if they dial 0 or timeout, ring an analog ringer and have a pickup code to grab the line. I am not sure that I can even do directed pickup on asterisk 1.2.13 Option 3 (I believe this is best, but am not sure where to start) When asterisk is in night mode, List all options and then if they dial 0 or timeout, park the call, then overhead page a record stating that there is a call that needs to be picked up on line 5401. They already have a button (with BLF) for 5401. How are others handling night calls when there is no receptionist available. Thank You, Steven BerkHolz Board member of Connectech Greater Detroit www.connectech.org Please visit us on the web at www.hirotecamerica.com HIROTEC AMERICA Ph. 248-836-5100 Fx. 248-836-5101 This e-mail and any files transmitted with it are intended only for the person or entity to which it is addressed and may contain confidential material and/or material protected by law. Any retransmission or use of this information may be a violation of that law. If you received this in error, please contact the sender and delete the material. ___ --Bandwidth and Colocation Provided 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] Realtime: Should I say or should I go (now) ?
Terry Wilson wrote: And a free hint: if you are going to have to do anything that resembles number porting, swapping extensions, etc.--don't use extensions/phone numbers as SIP usernames. You have to regenerate config files, etc. Make your SIP usernames meaningless and use func_odbc to look up what extension is tied to which device. I second that emotion. I consult with a bunch of people who rolled their own Asterisk systems long ago, and when the try to virtualize their system in various ways they find their hands are tied. It is indescribably confusing once the number in sip.conf gets disengaged from the extensions in the dialplan. I wouldn't say to make the names meaningless, though; there are different ways to use those names so that they have useful meaning. Just don't make them extension numbers; it's like the TCP/IP boundary between layers. See SIP for an example of the problems such a thing can cause :-) B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided 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] How to change sendmail return path
Yes did all that. I've configured sendmail before so know the basics. I tried modifying sendmail.mc and creating the sendmail.cf file and also tried modifying sendmail.cf directly. I always restart sendmail after changes. Would I need to create a noreply mailbox in sendmail perhaps? What creates the asterisk mailbox? Does that happen when I make asterisk? Maybe there are some clues in that script somewhere. -Original Message- From: Mojo with Horan Company, LLC [mailto:[EMAIL PROTECTED] Sent: Thursday, December 20, 2007 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to change sendmail return path Tilghman Lesher wrote: On Wednesday 19 December 2007 17:44:15 shadowym wrote: I had high hopes for this solution for unfortunately it's not working. Did exactly as you specified but return path is still [EMAIL PROTECTED] even though [EMAIL PROTECTED] in voicemail.conf :( Did you restart Sendmail? It doesn't pick up changes to its config file otherwise. And if you modified sendmail.mc instead of sendmail.cf, don't forget to regenerate sendmail.cf -- something like the following: cd /etc/mail; cp sendmail.cf sendmail.cf.todaysdate; m4 sendmail.mc sendmail.cf should work, followed by /etc/init.d/sendmail restart Mojo ___ --Bandwidth and Colocation Provided 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 330 beep on new VM
Steve Johnson wrote: This is pretty easy to suppress using the configuration files. Check: http://www.voip-info.org/wiki/view/Polycom+SoundPoint+IP+MWI+audio Looks easy once you have the config file provisioning in place, but it looks overkill and a lot of work to set this up for the only polycom phone I have (home). Regards, Ugo ___ --Bandwidth and Colocation Provided 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] 7970 CTLFile.tlv?
make sure the firmware in asterisk and the firmware on your phone matchup. it seems like your phone is trying to update it's firmware because the firmware on asterisk and your phone is different. Matthew Rubenstein [EMAIL PROTECTED] wrote: I've got in.atftpd running out of inetd: - /etc/inetd.conf tftpdgram udp waitnobody /usr/sbin/tcpd /usr/sbin/in.tftpd --logfile /tmp/atftpd.log --pidfile /tmp/atftpd.pid --tftpd-timeout 300 --retry-timeout 5 --mcast-port 1758 --mcast-addr 239.255.0.0-255 --maxthread 100 --verbose=5 --no-blksize /tftpboot --- But even when I run use a tftp client from a host on the inside network to retrieve the SEPMAC.cnf.xml file successfully, the /tmp/atftpd.log file is never touched, nor is the /tmp/atftpd.pid ever created. Even if I (touch /tmp/atftpd.log; chown nobody.nogroup /tmp/atftpd.log) the status files are untouched. But I am getting the requested file. Also, what do I do to use an XmlDefault.cnf.xml file? Just rename the SEP.cnf.xml file to that? I also saw on the Web someone who had my problem with the 7970, but cryptically noted that they solved their problem which was wrong platform newline terminations. What chars does the 7970 need for its conf files newlines to be? On Fri, 2007-12-21 at 09:47 -0600, Jason Parker wrote: You don't need the .tlv file. It's optional, and will be skipped if it cannot be found. Your problem is elsewhere. I've found that the 7970s are very finicky. I've never had luck with the SEPMAC.cnf.xml - only XmlDefault.cnf.xml (case may vary - check your tftp logs) Matthew Rubenstein wrote: I've got a Cisco 7970 that's not completing its network registration to Asterisk. The Registering message stays on the screen (with the moving time wheel). After a few minutes, the onscreen message flashes Updating CTL then Loading..., then the status messages update with: No valid CAPF server File Not Found: CTLFile.tlv No CTL installed SEPMAC.cnf.xml (where MAC is the phone's MAC addr minus :s) before repeating the cycle (forever). Where can I get a CTLFile.tlv , or remove the requirement for it? Or is there another way to fix this problem? TIA. Asterisk v1.2.9.1-BRIstuffed-0.3.0-PRE-1q SCCP firmware Load File: TERM70.7-0-1-0s App Load ID: Jar70.2-9-0-117.sbn JVM Load ID: CVM70.2-0-0-112.sbn OS Load ID: cnu70.2-7-4-134.sbn Boot Load ID: 7970_64060118.bin -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided 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] Upgrade to Asterisk 1.4 - it's one year's old!
On Friday 21 December 2007 13:16:17 Matt wrote: It may be a year old.. but until Digium is drinking their own dog food.. I won't be using it. I beg your pardon. The Digium IVR has been on 1.4 since about April or so. -- 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] SIP hangup on call proceeding message
Has anyone experienced the situation where you receive a PRI_EVENT_PROGRESS message from a PRI that is then sent to a SIP channel where the SIP client (tried 2 different phones/manufactures) never acknowledges, Asterisk resends the message two more time and then begins hanging the call up? This is happening to me when my long distance carrier turns on account codes (you make a long distance call that is routed to the carrier, they send the caller a tone, then the caller enters in an account code for billing purpose). The call will connect, you can talk to the called party for 20 seconds and then the call drops. Thanks for any 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] best way for night ringer??
BerkHolz, Steven wrote: Option 3 (I believe this is best, but am not sure where to start) When asterisk is in night mode, I'm doing option 3, menu item on the IVR to ring the night bell. Plays an awfully loud horn noise on the PA while ringing a phone out an the plant floor every 15 seconds. The breakroom and plant manager's phones are apart of the same pickup group. Any of them can do a *7 to grab an incoming call. exten = 4173,1,GotoIfTime(07:45-17:00|mon-fri|*|*?press-officehours,s,1) exten = 4173,n,System(/bin/cp /usr/local/bin/bullhorn.call /var/spool/asterisk/outgoing/bullhorn`date +%s`.call) exten = 4173,n,Dial(SIP/4173,15,tT) exten = 4173,n,Goto(analog-extensions,4173,1) Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] best way for night ringer??
I have mine set up to ring a group of designated phones. Each one of those phones has a dedicated line button that subscribes to their particular account in the group. This way when the phone rings the user KNOWS that it is the main building number that is ringing. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BerkHolz, Steven Sent: Friday, December 21, 2007 2:29 PM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] best way for night ringer?? Asterisk 1.2.13 I am trying to figure out the best way for a night bell at work. Note: I have no spare buttons available on the phones. But I do have two lines and two park positions as buttons. Option 1 (easiest and the one I just implemented) When asterisk is in night mode, Connect to IVR, List all options and then if they dial 0 or timeout, ring every phone in the building. Option 2 (This would require user training) When asterisk is in night mode, List all options and then if they dial 0 or timeout, ring an analog ringer and have a pickup code to grab the line. I am not sure that I can even do directed pickup on asterisk 1.2.13 Option 3 (I believe this is best, but am not sure where to start) When asterisk is in night mode, List all options and then if they dial 0 or timeout, park the call, then overhead page a record stating that there is a call that needs to be picked up on line 5401. They already have a button (with BLF) for 5401. How are others handling night calls when there is no receptionist available. Thank You, Steven BerkHolz Board member of Connectech Greater Detroit www.connectech.org Please visit us on the web at www.hirotecamerica.com HIROTEC AMERICA Ph. 248-836-5100 Fx. 248-836-5101 This e-mail and any files transmitted with it are intended only for the person or entity to which it is addressed and may contain confidential material and/or material protected by law. Any retransmission or use of this information may be a violation of that law. If you received this in error, please contact the sender and delete the material. ___ --Bandwidth and Colocation Provided 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] best way for night ringer??
Doug Lytle wrote: BerkHolz, Steven wrote: Option 3 (I believe this is best, but am not sure where to start) When asterisk is in night mode, I'm doing option 3, menu item on the IVR to ring the night bell. Plays an awfully loud horn noise on the PA while ringing a phone out an the plant floor every 15 seconds. The breakroom and plant manager's phones are apart of the same pickup group. Any of them can do a *7 to grab an incoming call. exten = 4173,1,GotoIfTime(07:45-17:00|mon-fri|*|*?press-officehours,s,1) exten = 4173,n,System(/bin/cp /usr/local/bin/bullhorn.call /var/spool/asterisk/outgoing/bullhorn`date +%s`.call) exten = 4173,n,Dial(SIP/4173,15,tT) exten = 4173,n,Goto(analog-extensions,4173,1) Doug Dough, I see that you use cp to copy the call file to spool directory, that is not recommended, use mv instead since it is a atomic command whitch cp isnt. So an if you change it to : exten = 4173,1,GotoIfTime(07:45-17:00|mon-fri|*|*?press-officehours,s,1) exten = 4173,n,System(/bin/cp /usr/local/bin/bullhorn.call /temp/bullhorn.call) exten = 4173,n,System(/bin/mv /temp/bullhorn.call /var/spool/asterisk/outgoing/bullhorn`date +%s`.call) exten = 4173,n,Dial(SIP/4173,15,tT) exten = 4173,n,Goto(analog-extensions,4173,1) should solve it. /Mats ___ --Bandwidth and Colocation Provided 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] Send SIP 100 Trying instead of 183 Session Progress
You are probably running into the problem described below. Below that is a link to the original document with the code patch. I put it on a PRI box we use inhouse and it took care of the 183 before a busy for me. However, this is a box we use inhouse. I've never put it on anything in production. Your mileage may vary gday guys (n'gals). I have a third party SIP platform which generates outbound calls via asterisk to ISDN (Australia - so thats ETSI ISDN). This platform doesn't really like inband signalling on outbound calls (ie getting 183's with SDP -- its fine with 180 Ringing etc...) Having had a bit of a silly time with the sip.conf variable progressinband=never,no,yes (arg!) I dug a little deeper into the chan_sip code. It appears on a SIP-Zap call the ISDN channel is opened, and before you can say 'boo' sip_write() in chan_sip is called this appears to occurs prior to any ISDN signalling (such as PRI_EVENT_PROCEEDING etc..) sip_write doesn't seem to care at all what progressinband is set to, and if it gets a frame when the SIP channel is not in AST_STATE_UP it generates a 183 with SDP (then sets SIP_PROGRESS_SENT) Does this behaviour seem strange? I'm not really sure if this is a bug, a 'its just like that' thing, or something strange with our ISDN that is unusual? In an ideal world (for me anyway... *grin*) I would think that progressinband=never (or even progressinband=no) would mean that 180 Ringing, 486 Busy etc would be used and 183 Session Progress with SDP would not... I have done some basic testing and if I patch as follows... url to patch document: From ds at seiros.ru Mon May 1 04:41:40 2006 From: ds at seiros.ru ... Richard On Dec 21, 2007, at 9:57 AM, Remi Quezada wrote: Hi, I have a Asterisk that connects to the PSTN via a PRI. After Asterisk sends the setup message it immediately sends a 183 Session Progress. Is there a way I can change it so that it sends a 100 Trying instead? Because I am having some issues with a equipment where it does not play a busy tone as a result of sending a 183 Session Progress then the 486 Busy. Thanks Remi ___ --Bandwidth and Colocation Provided 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] Realtime: Should I say or should I go (now) ?
On Dec 21, 2007, at 1:29 PM, Brian Capouch wrote: Terry Wilson wrote: config files, etc. Make your SIP usernames meaningless and use func_odbc to look up what extension is tied to which device. I wouldn't say to make the names meaningless, though; there are different ways to use those names so that they have useful meaning. Just don't make them extension numbers; it's like the TCP/IP boundary between layers. See SIP for an example of the problems such a thing can cause :-) Since we were a nationwide VoIP provider and provisioned phones and had to deal with returns, etc. The SIP username really couldn't be used to identify anything. Since they were multi-line phones, etc. the only way we could have had a meaningful username would be possibly MACaddress-LineNumber or something. But if you have a normalized database, looking up user from SIP account becomes trivial. :-) ___ --Bandwidth and Colocation Provided 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] Send SIP 100 Trying instead of 183 Session Progress
And in case that link doesn't work so well in text email clients here is the real address. lists.digium.com/pipermail/asterisk-dev/2006-May.txt.gz Richard On Dec 21, 2007, at 4:24 PM, Richard Revels wrote: You are probably running into the problem described below. Below that is a link to the original document with the code patch. I put it on a PRI box we use inhouse and it took care of the 183 before a busy for me. However, this is a box we use inhouse. I've never put it on anything in production. Your mileage may vary gday guys (n'gals). I have a third party SIP platform which generates outbound calls via asterisk to ISDN (Australia - so thats ETSI ISDN). This platform doesn't really like inband signalling on outbound calls (ie getting 183's with SDP -- its fine with 180 Ringing etc...) Having had a bit of a silly time with the sip.conf variable progressinband=never,no,yes (arg!) I dug a little deeper into the chan_sip code. It appears on a SIP-Zap call the ISDN channel is opened, and before you can say 'boo' sip_write() in chan_sip is called this appears to occurs prior to any ISDN signalling (such as PRI_EVENT_PROCEEDING etc..) sip_write doesn't seem to care at all what progressinband is set to, and if it gets a frame when the SIP channel is not in AST_STATE_UP it generates a 183 with SDP (then sets SIP_PROGRESS_SENT) Does this behaviour seem strange? I'm not really sure if this is a bug, a 'its just like that' thing, or something strange with our ISDN that is unusual? In an ideal world (for me anyway... *grin*) I would think that progressinband=never (or even progressinband=no) would mean that 180 Ringing, 486 Busy etc would be used and 183 Session Progress with SDP would not... I have done some basic testing and if I patch as follows... url to patch document: From ds at seiros.ru Mon May 1 04:41:40 2006 From: ds at seiros.ru ... Richard On Dec 21, 2007, at 9:57 AM, Remi Quezada wrote: Hi, I have a Asterisk that connects to the PSTN via a PRI. After Asterisk sends the setup message it immediately sends a 183 Session Progress. Is there a way I can change it so that it sends a 100 Trying instead? Because I am having some issues with a equipment where it does not play a busy tone as a result of sending a 183 Session Progress then the 486 Busy. Thanks Remi ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to change sendmail return path
shadowym wrote: What creates the asterisk mailbox? Does that happen when I make asterisk? The return-path is set to [EMAIL PROTECTED] Your asterisk process is running as the user 'asterisk', no? If so, that's why you're getting the return path set to 'asterisk'. I haven't used sendmail in about a decade, but (brain's digging deep into old memories now) I think you need to call sendmail with the -f option. Just checked the man page (man sendmail - you *did* look here, didn't you? ;) and came up with: -fname Sets the name of the ''from'' person (i.e., the envelope sender of the mail). This address may also be used in the From: header if that header is missing during initial submission. The envelope sender address is used as the recipient for delivery status notifications and may also appear in a Return-Path: header. -f should only be used by ''trusted'' users (normally root, daemon, and network) or if the person you are trying to become is the same as the person you are. Otherwise, an X-Authentication-Warning header will be added to the message. So there you have it. You not only need to add the asterisk user to the trusted users list, but you also need to make sure that whenever sendmail is being invoked, it's called with the -f option. I don't know if asterisk does this or not - perhaps others could confirm whether it does or doesn't, and where/if this can be changed. HTH, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] best way for night ringer??
MatsK wrote: Doug, I see that you use cp to copy the call file to spool directory, that is not recommended, use mv instead since it is a atomic command whitch cp isnt. Thanks! I'll have to update that. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime: Should I say or should I go (now) ?
And a free hint: if you are going to have to do anything that resembles number porting, swapping extensions, etc.--don't use extensions/phone numbers as SIP usernames. You have to regenerate config files, etc. Make your SIP usernames meaningless and use func_odbc to look up what extension is tied to which device. I second that emotion. I consult with a bunch of people who rolled their own Asterisk systems long ago, and when the try to virtualize their system in various ways they find their hands are tied. It is indescribably confusing once the number in sip.conf gets disengaged from the extensions in the dialplan. I wouldn't say to make the names meaningless, though; there are different ways to use those names so that they have useful meaning. Just don't make them extension numbers; it's like the TCP/IP boundary between layers. See SIP for an example of the problems such a thing can cause :-) B. Within my Realtime Asterisk Cluster, I use Directory Numbers (DN) for all sip/iax devices. These are a 5 or 6 digit number that don't mean a whole lot until I assign an extension to it in the dial plan. So DN 22331 could be exten 101 or exten 1001 and can be updated or changed to a different extension in the dial plan without having to update the device itself, unless the CID needs to be changed. You need very good record keeping to be successful. Also on the phone device, the auth name or account name may be 22331 but the display name will be 1001. To make this change, you need a central provisioning server, update the config file and reboot the phone to update the display name. Hope this helps and doesn't confuse things. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided 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] ODBC Voicemail and performance....
Running on branch/1.4 I have been watching some the queries from Asterisk and I think I have a place where some efficiency can come, but I am at a lost as to what is calling it... It appears that every 10-15 seconds, every mailbox INBOX and Old gets queried for the number of voice mail files. I have exposed SIP to verify that it wasn't the phones requesting. It is not much of a problem in a small voice mail system, but if there are 500 mailboxes, that is over 1000 queries per event and if it is occurring every 15 seconds that is 4000 queries a minute just to see if the voice mail has change. So I would like to change this, just not sure where to look. It seems to me we only need to query this when we call the VM app or a user enters the VM system. That does leave a problem for outside changes via the database on clustered systems, but... I believe there is a way around that where we could have 1 query every defined number of seconds. Just need to know how to find out what is polling that function. ___ --Bandwidth and Colocation Provided 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] Control playback
Hi All I have been asked if it is possible for an external application to be aware of the position of the playbcak of a file with control playback ie a file is playing and the user hits the fast forward button , is there a manager event that show how far into the file it has been played? thanks in advance Robb ___ --Bandwidth and Colocation Provided 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] ODBC Voicemail and performance....
So I would like to change this, just not sure where to look. It seems to me we only need to query this when we call the VM app or a user enters the VM system. Maybe the same code that calls the externnotify command (a custom post-exec script) could trigger the SIP notification beforehand. A nice event-based solution. :) Additionally it would have to be triggered whenever a user (/peer/phone, whatever) registers ... Did not really think it through. Regards, Philipp Kempgen I was even thinking to just create a single SELECT COUNT(*) FROM voicemail with no parameters and store the value. This way if something changed, you could at least then update the voicemail and notify the users without a event triggering it. You could have this run ever minute or so (or less by configuration) but at least the SQL query would be singular and not for every WHERE clause available and improve the overall resources on the SQL and Asterisk boxes. ___ --Bandwidth and Colocation Provided 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] ODBC Voicemail and performance....
Tony Plack wrote: Running on branch/1.4 I have been watching some the queries from Asterisk and I think I have a place where some efficiency can come, but I am at a lost as to what is calling it... It appears that every 10-15 seconds, every mailbox INBOX and Old gets queried for the number of voice mail files. I have exposed SIP to verify that it wasn't the phones requesting. It is not much of a problem in a small voice mail system, but if there are 500 mailboxes, that is over 1000 queries per event and if it is occurring every 15 seconds that is 4000 queries a minute just to see if the voice mail has change. So I would like to change this, just not sure where to look. It seems to me we only need to query this when we call the VM app or a user enters the VM system. Maybe the same code that calls the externnotify command (a custom post-exec script) could trigger the SIP notification beforehand. A nice event-based solution. :) Additionally it would have to be triggered whenever a user (/peer/phone, whatever) registers ... Did not really think it through. 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] Polycom 330 beep on new VM
Ugo Bellavance wrote: Steve Johnson wrote: This is pretty easy to suppress using the configuration files. Check: http://www.voip-info.org/wiki/view/Polycom+SoundPoint+IP+MWI+audio Looks easy once you have the config file provisioning in place, but it looks overkill and a lot of work to set this up for the only polycom phone I have (home). Regards, Ugo ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users That may very well be, but I can't find any control over that in the phone's web interface. I had to get dirty with the xml files, but it wasn't bad at all. Aside from the CONTENT of the files, I only needed to create a user on my linux box and make sure there was an ftp daemon installed at bare minimum. the ftp server address can then be input into the phone along with the username and password. Then, for the content, I found some excellent samples on the web that needed next to no configuration to be useful to me. Come to think of it, I am pretty sure I started with the ones at http://www.krisk.org/asterisk/pcom/ Mojo ___ --Bandwidth and Colocation Provided 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] Dead Incoming call - Sangoma A200
:48 VERBOSE[15840] logger.c: -- Executing NoOp(Zap/2-1, Using CallerID ) in new stack Dec 21 13:43:48 DEBUG[15840] app_macro.c: Executed application: NoOp Dec 21 13:43:48 DEBUG[15840] pbx.c: Expression result is '1' Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing GotoIf(Zap/2-1, 1?skipdb) in new stack Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Goto (ext-group,600,4) Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing Set(Zap/2-1, __NODEST=) in new stack Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing Set(Zap/2-1, __BLKVM_OVERRIDE=BLKVM/600/Zap/2-1) in new stack Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing Set(Zap/2-1, __BLKVM_BASE=600) in new stack Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing Set(Zap/2-1, DB(BLKVM/600/Zap/2-1)=TRUE) in new stack Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing Set(Zap/2-1, RRNODEST=) in new stack Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing Set(Zap/2-1, __NODEST=600) in new stack Dec 21 13:43:48 DEBUG[15840] pbx.c: Expression result is '1' Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing GotoIf(Zap/2-1, 1?REPCID) in new stack Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Goto (ext-group,600,15) Dec 21 13:43:48 DEBUG[15840] pbx.c: Function result is '' Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing NoOp(Zap/2-1, CALLERID(name) is ) in new stack Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing Set(Zap/2-1, _RGPREFIX=Leongatha:) in new stack Dec 21 13:43:48 DEBUG[15840] pbx.c: Function result is '' Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing Set(Zap/2-1, CALLERID(name)=Leongatha:) in new stack Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing Set(Zap/2-1, RecordMethod=Group) in new stack Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing Macro(Zap/2-1, record-enable|203-202-200-201-208-209-206|Group) in new stack Dec 21 13:43:48 DEBUG[15840] pbx.c: Function result is '0' Dec 21 13:43:48 DEBUG[15840] pbx.c: Expression result is '0' Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing GotoIf(Zap/2-1, 0?2:4) in new stack Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Goto (macro-record-enable,s,4) Dec 21 13:43:48 DEBUG[15840] app_macro.c: Executed application: GotoIf Dec 21 13:43:48 DEBUG[15840] pbx.c: Function result is '20071221-134348' Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing AGI(Zap/2-1, recordingcheck|20071221-134348|1198205023.1657) in new stack Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck Dec 21 13:43:48 VERBOSE[15840] logger.c: -- AGI Script recordingcheck completed, returning 0 Dec 21 13:43:48 DEBUG[15840] app_macro.c: Executed application: AGI Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing NoOp(Zap/2-1, No recording needed) in new stack Dec 21 13:43:48 DEBUG[15840] app_macro.c: Executed application: Noop Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing Set(Zap/2-1, RingGroupMethod=hunt) in new stack Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing Macro(Zap/2-1, dial|20||203-202-200-201-208-209-206) in new stack Dec 21 13:43:48 DEBUG[15840] pbx.c: Expression result is '1' Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing GotoIf(Zap/2-1, 1?dial) in new stack Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Goto (macro-dial,s,3) Dec 21 13:43:48 DEBUG[15840] app_macro.c: Executed application: GotoIf Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing AGI(Zap/2-1, dialparties.agi) in new stack Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi Dec 21 13:43:48 VERBOSE[15840] logger.c: dialparties.agi: Starting New Dialparties.agi Dec 21 13:43:48 DEBUG[15847] manager.c: Manager received command 'login' Dec 21 13:43:48 VERBOSE[15847] logger.c: == Parsing '/etc/asterisk/manager.conf': Dec 21 13:43:48 VERBOSE[15847] logger.c: == Parsing '/etc/asterisk/manager.conf': Found Dec 21 13:43:48 VERBOSE[15847] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': Dec 21 13:43:48 VERBOSE[15847] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': Found Dec 21 13:43:48 DEBUG[15847] acl.c: 0.0.0.0/0.0.0.0/0.0.0.0 appended to acl for peer Dec 21 13:43:48 DEBUG[15847] acl.c: 127.0.0.1/255.255.255.0/255.255.255.0 appended to acl for peer Dec 21 13:43:48 DEBUG[15847] acl.c: # Testing 127.0.0.1 with 0.0.0.0 Dec 21 13:43:48 DEBUG[15847] acl.c: # Testing 127.0.0.1 with 127.0.0.0 Dec 21 13:43:48 VERBOSE[15847] logger.c: == Manager 'admin' logged on from 127.0.0.1 Dec 21 13:43:48 VERBOSE[15840] logger.c: dialparties.agi: Caller ID name is 'Leongatha:' number is 'unknown' Dec 21 13:43:48 VERBOSE[15840] logger.c: dialparties.agi: USE_CONFIRMATION: 'FALSE' Dec 21 13:43:48 VERBOSE[15840] logger.c: dialparties.agi: RINGGROUP_INDEX: '' Dec 21 13:43:48 VERBOSE[15840] logger.c: dialparties.agi
[asterisk-users] On-the-phone
Hi there, I have a Polycom phone that has two extensions registered to it, let's say 200 201. Is there anyway to code in the Asterisk dialplan to show BOTH lines are busy when either of 200 or 201 are in use? Reason is that my Polycom phones will show the presence info via BLF red light, but I would have to have 2 separate entries in the monitoring phone (one for each extension), even though it's only one device. 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
[asterisk-users] call-limit in database
hi, all proble: I have add CALL-LIMIT field in my sip table in mysql. but when i call using sip same error occurred when use simple sip.conf file. is this possible to add CALL-LIMIT field in sip realtime table in mysql. if yes than how Bhrugu Mehta ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call-limit in database
I will be out of the office until Wednesday, January 2, 2008. If this is an emergency, please call Customer Service at (877) 791-7700. Thank you have a great holiday season! ___ --Bandwidth and Colocation Provided 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] Upgrade to Asterisk 1.4 - it's one year's old!
21 dec 2007 kl. 10.12 skrev Remco Barendse: I wonder if there are any major obstacles for upgrading. Just tried an in-place upgrade on my home box : make[1]: Leaving directory `/usr/src/asterisk-addons-1.4.5' for x in app_addon_sql_mysql.so app_saycountpl.so cdr_addon_mysql.so res_config_mysql.so; do /usr/bin/install -c -m 755 $x /usr/lib/asterisk/modules ; done /usr/bin/install: cannot stat `app_addon_sql_mysql.so': No such file or directory /usr/bin/install: cannot stat `cdr_addon_mysql.so': No such file or directory /usr/bin/install: cannot stat `res_config_mysql.so': No such file or directory make: *** [install] Error 1 For some reason, the mysql modules wasn't compiled. Did you check the requirements for mysql and read the compile errors? It's not shown here. And the asterisk console is flooded with these errors : [Dec 21 10:10:58] WARNING[22897]: chan_sip.c:6707 determine_firstline_parts: Bad request protocol Packet [Dec 21 10:10:58] WARNING[22897]: chan_sip.c:6707 determine_firstline_parts: Bad request protocol Packet So for the next time to come i'll turn back to 1.2 :) The chan_sip messages was only warnings, nothing serious. Propably strange NAT Keepalives, like those I've seen from cirpak devices. Communication should work as expected. If you give up for these errors, you might consider buying Asterisk Business Edition where everything is precompiled and easy-to-install, and you have support. Thanks for the feedback! Best regards, /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
Re: [asterisk-users] Send SIP 100 Trying instead of 183 Session Progress
21 dec 2007 kl. 22.24 skrev Richard Revels: You are probably running into the problem described below. Below that is a link to the original document with the code patch. I put it on a PRI box we use inhouse and it took care of the 183 before a busy for me. However, this is a box we use inhouse. I've never put it on anything in production. Your mileage may vary gday guys (n'gals). I have a third party SIP platform which generates outbound calls via asterisk to ISDN (Australia - so thats ETSI ISDN). This platform doesn't really like inband signalling on outbound calls (ie getting 183's with SDP -- its fine with 180 Ringing etc...) Having had a bit of a silly time with the sip.conf variable progressinband=never,no,yes (arg!) I dug a little deeper into the chan_sip code. It appears on a SIP-Zap call the ISDN channel is opened, and before you can say 'boo' sip_write() in chan_sip is called this appears to occurs prior to any ISDN signalling (such as PRI_EVENT_PROCEEDING etc..) sip_write doesn't seem to care at all what progressinband is set to, and if it gets a frame when the SIP channel is not in AST_STATE_UP it generates a 183 with SDP (then sets SIP_PROGRESS_SENT) Does this behaviour seem strange? I'm not really sure if this is a bug, a 'its just like that' thing, or something strange with our ISDN that is unusual? In an ideal world (for me anyway... *grin*) I would think that progressinband=never (or even progressinband=no) would mean that 180 Ringing, 486 Busy etc would be used and 183 Session Progress with SDP would not... I don't think progressinband controls early media (audio to caller before call setup) but how indications should be sent (in audio=inband). If we get early media from the callee leg of the call, we have to relay it always. If you get early media signalling in SIP and don't have early media on the outbound call leg, then there's a bug and you should open a bug in the bug tracker so we can resolve it. For license reasons, we can't handle patches on the mailing list, we have to get them through the bug tracker. I really appreciate your help in resolving this issue, as you clearly have a lot of insight in the situation. Please open a bug on the bug tracker and we'll meet you there! Thanks, /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ * The Asterisk SIP Masterclass - Stockholm, Sweden, January 2008 * Register today! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call-limit in database
22 dec 2007 kl. 06.40 skrev Bhrugu Mehta: hi, all proble: I have add CALL-LIMIT field in my sip table in mysql. but when i call using sip same error occurred when use simple sip.conf file. You can check if it works with sip show peer. The call limit you set in the database should be visible there. is this possible to add CALL-LIMIT field in sip realtime table in mysql. Absolutely. What is your problem, you did not specify that. /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
[asterisk-users] Summary: Upgrading to Asterisk 1.4
Friends, Thanks for all the feedback. If you have additional success stories or important issues, feel free to continue the discussion. I've learned a lot from your input. As a developer, I spend too much time in the bug tracker, working with particular bugs, so I often wonder how on earth anyone can use this buggy platform for anything business-like. It really feels good to get reports on people successfully using our software and meet Asterisk users who just love the product and handle tons of calls every hour with it. And as a developer, everything is of course more simple and you live in the future, moving forward to new features, new functions all the time based on customer requirements or feature requests in the mailing list or the bug tracker... Now over to a summary of the feedback. I'm not going deeper into bugs reported, those will be handled separately. * DON'T TOUCH MY ASTERISK PBX This discussion about the 1.4 upgrade situation has given very important feedback. First, for a lot of users there's simply no reason to upgrade a PBX everytime we release a new Asterisk. Existing installations that work should not be touched unless there's a very good reason to, like a new feature that makes business sense. Just upgrading for the cause of upgrading is a feature of the non-open software industry that gets a lot of revenue from upgrades. We developers has to accept that people appreciate our work, but decide not to upgrade every installation at every release. We might have to reconsider our support policy here, where we developers abandoned 1.2 this summer. We might need another team that runs 1.2 support in the bug tracker. * MAKE UPGRADING EASIER Another issue is to make the upgrade much smoother. We can't anticipate that people upgrade from 1.0 to 1.2 to 1.4 and read all the docs for every release. They can jump from 0.8 to 1.4. Or 1.0 to the future release of 1.6. We need to assist that and haven't made a good effort in doing so. But even for upgrades from 1.2 to 1.4, we need to be more clear about changes that are required, especially for 1.2 installations that already was upgraded from 1.0 and still use the 1.0 configuration syntax. They are going to have a broken configuration in 1.4 and this is the first time that happens in Asterisk. We need to make clear that Asterisk admins need to go through the log files in 1.2 and check all deprecation warnings. These needs to be fixed before even testing 1.4. * USE ASTERISK 1.4 FOR NEW INSTALLATIONS, PLEASE My personal goal would be to get the community to start using 1.4 for all new installations. We need to produce information to help this upgrade path. It's not about upgrading systems, since we're talking about new installations. It's about upgrading the Asterisk admins and installers - human beings. The success stories reported to me personally and on the list indicates that 1.4 is indeed ready for production and it's a great product. With that, I'm now changing my focus from SIP invite states, RTP sessions and video formats to Christmas ham purchasing, baking Christmas bread (julvört) and decorating the Christmas tree. Of course, you understand that there's an Asterisk asterisk on top of all those trees, right? :-) After Christmas, I'm running the new Asterisk SIP Masterclass together with Daniel Mierla here in Stockholm. He's one of the core OpenSER developers and it's going to be a great class. I'm sure we will locate a set of new interesting bugs in svn trunk during that week. I'm really looking forward to that training. (Hint: We still have a few open seats... :-) ) Greetings from a dark and cold place in Sweden, without a decent amount of snow... Have a wonderful, merry and cheerful Christmas! /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
