Re: [Asterisk-Users] Music On Hold and ATA-186 w/Silence Supression
Alejandro G wrote: I have a problem with ATA-186 configured for silence supression (AudioMode bit 0 = 1). When enabled and listening music on hold no sound is heared (if I talk I began to hear the music and again mutes when I stop talking). If I configure for silence supression off everything goes fine. Any hint? Don't use silence supression. Asterisk doesn't support it. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to configure groups using a sip phone
hi friends ! i am facing a problem from one week and now required ur help urgently. Actually, i want to configure asterisk for two groups javgroup and linuxgroup. i also have constraint to use only sip phone (esatara ). now, please help me is it possible to configure astersik in that way or that kind of facility is given in zapata.conf. tell me in detail abt the configurations of the sip.conf and extensions.conf. thanks Deepak Dhiman ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to configure groups using a sip phone
Can you be more specific? What are you trying to achieve with the creation of such groups? - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, April 04, 2005 3:50 PM Subject: [Asterisk-Users] how to configure groups using a sip phone hi friends ! i am facing a problem from one week and now required ur help urgently. Actually, i want to configure asterisk for two groups javgroup and linuxgroup. i also have constraint to use only sip phone (esatara ). now, please help me is it possible to configure astersik in that way or that kind of facility is given in zapata.conf. tell me in detail abt the configurations of the sip.conf and extensions.conf. thanks Deepak Dhiman ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Realtime Capabilities
The term RTCache has never been mentioned in the WIKI or these forums. I assume that it's some sort of function to speed up realtime db access by keeping transactions in RAM and writing periodically? If so, I can understand why this would need to be flushed. - Original Message - From: Matthew Boehm [EMAIL PROTECTED] To: Asterisk Users asterisk-users@lists.digium.com Sent: Monday, April 04, 2005 3:01 PM Subject: Re: [Asterisk-Users] Asterisk Realtime Capabilities to a load-balanced (not sure which mechanism I'll empoy here yet) I was quoted a $21,000 layer-7 switch from F5 Networks to do SIP load balancing. outside)? In other words, can the registering server update a USRLOC type database on the fly, so all other servers know where to route calls As long as all * servers share the same central database; this way when SIP 1 registers via RealTime at server A, server B (using same db) should be able to see the registration. You may not be able to use RTCache though... Also, I will be using multiple * boxes as media gateways. Is there an existing mechanism whereby a given server can record the number of busy/available ZAP channels to a central database for the purpose of call routing? Nothing built-in comes to mind, but Im sure you could AGI something. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Previous sip reload not yet done
Hi list, We are running a CVS version of 03-30-2005 but also had this behaviour on previous versions. From time to time, after a period of not making calls (eg a night or few hours), we have no dialtone when we want to call. SIP show peers show EP registered with status OK but nothing happend. Nothing special in logs. After a SIP reload, everything is again working fine. So we add a SIP reload each morning in crontab. But this is not solving the problem: it's not always efficient and when we try to re-run this command from CLI, we get a Previous sip reload not yet done. Only solution is to restart asterisk. Does anyone else have this problem? Is there a workaround? Thanks for any hints. -- Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: AS5300+SIP+ASTERISK or AS5300+MGCP
AS5300 setup =~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2005.04.04 09:37:31 =~=~=~=~=~=~=~=~=~=~=~= sh runn Building configuration... Current configuration : 11599 bytes ! ! Last configuration change at 03:26:25 GMT Mon Apr 4 2005 by charles ! NVRAM config last updated at 03:06:50 GMT Mon Apr 4 2005 by charles ! version 12.3 service timestamps debug datetime localtime show-timezone service timestamps log datetime localtime show-timezone service password-encryption ! hostname 66.178.100.66 ! boot-start-marker boot-end-marker ! ! ! resource-pool enable --More-- ! resource-pool throttle 20 default clock timezone GMT 3 clock calendar-valid ! aaa new-model ! ! aaa group server radius cdt-1 server 159.148.8.108 auth-port 2362 acct-port 2363 ! aaa group server radius cdt-2 server 62.85.77.82 auth-port 2362 acct-port 2363 ! aaa group server radius tsl server 62.56.250.200 auth-port 1812 acct-port 1813 ! aaa authentication login h323 group radius aaa accounting send stop-record authentication failure aaa accounting connection h323 stop-only broadcast group cdt-1 group cdt-2 group tsl aaa nas port voip aaa session-id common --More-- ip subnet-zero ip telnet source-interface FastEthernet0 ip name-server 66.178.100.68 ! ! ! trunk group mgcp ! isdn switch-type primary-net5 ! voice rtp send-recv ! voice service pots ! voice service voip cause-code legacy h323 h225 timeout setup 8 session transport udp sip min-se 600 ! voice class codec 2 --More-- codec preference 1 gsmfr ! ! voice class permanent 1 signal timing idle suppress-voice 5 signal timing oos suppress-all 30 signal timing oos timeout 120 ! ! voice class h323 1 h225 timeout tcp establish 30 h225 timeout connect 60 h225 timeout setup 30 call start fast ! voice class h323 2 call start slow ! voice class h323 1001 call start fast ! voice class h323 10 ! --More-- ! voice class busyout 1 ! ! voice class dualtone-detect-params 1 ! ! ! ! ! fax interface-type modem ! ! controller E1 0 clock source line primary ds0-group 0 timeslots 1-15,17-31 type r2-digital ! ! ! translation-rule 22 Rule 0 22254 254 ! ! ! interface Tunnel1 ip address 192.168.44.1 255.255.255.0 tunnel source Ethernet0 tunnel destination 217.21.95.9 ! interface Tunnel17 --More-- ip address 10.1.17.2 255.255.255.0 shutdown tunnel source 212.165.147.254 tunnel destination 66.92.133.199 tunnel mode nos ! interface Tunnel18 no ip address ! interface Ethernet0 ip address 195.202.73.106 255.255.255.248 no ip mroute-cache ! interface Serial0 no ip address no ip mroute-cache clockrate 2015232 no fair-queue ! interface Serial1 no ip address no ip mroute-cache clockrate 2015232 --More-- no fair-queue ! interface Serial2 no ip address no ip mroute-cache clockrate 2015232 no fair-queue ! interface Serial3 no ip address no ip mroute-cache shutdown clockrate 2015232 fair-queue 100 256 0 ip rtp priority 1 1 75 ! interface Serial2:15 no ip address isdn switch-type primary-net5 no cdp enable ! interface FastEthernet0 ip address 172.16.202.90 255.255.255.0 secondary --More-- ip address 66.178.100.66 255.255.255.248 ip access-group 1 in ip access-group 1 out no ip mroute-cache duplex auto speed auto h323-gateway voip interface h323-gateway voip id gk0 ipaddr 216.52.153.203 1719 h323-gateway voip h323-id ngins ip rtp priority 16384 16383 400 ! ip classless ip route 0.0.0.0 0.0.0.0 66.178.100.65 no ip http server ! ! no logging trap access-list 101 permit ip any any ! route-map VOIP permit 20 match ip address 101 ! route-map VOIP permit 100 --More-- ! ! radius-server attribute 44 include-in-access-req radius-server host 159.148.8.108 auth-port 2362 acct-port 2363 key 7 065E582A585C51411F0317 radius-server host 62.85.77.82 auth-port 2362 acct-port 2363 key 7 014B510F4F195E573B584B radius-server host 62.56.250.200 auth-port 1812 acct-port 1813 key 7 121500031F0E050A radius-server retransmit 10 radius-server timeout 120 radius-server vsa send accounting radius-server vsa send authentication call threshold global total-calls low 60 high 90 busyout ! call application voice kenya flash:kenya.tcl ! call application voice kenya1 flash:kenya.tcl ! ! voice-port 0:0 compand-type a-law connection plar 9001 ! ! mgcp call-agent 62.56.250.198 2427 service-type mgcp version 1.0 mgcp dtmf-relay voip codec all mode out-of-band mgcp restart-delay 2 mgcp codec g711ulaw packetization-period 10 mgcp package-capability dtmf-package mgcp package-capability line-package mgcp package-capability rtp-package mgcp package-capability nas-package mgcp package-capability script-package mgcp sdp simple --More-- no mgcp validate domain-name mgcp endpoint offset mgcp bind control source-interface FastEthernet0 mgcp bind media source-interface FastEthernet0 mgcp behavior signals v0.1 ! mgcp profile default ! dial-peer cor custom ! ! ! ! dial-peer voice 271 pots permission orig huntstop
Re: [Asterisk-Users] V92 modem with asterisk
No. - Original Message - From: Alexandre Charles [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, April 04, 2005 3:48 PM Subject: [Asterisk-Users] V92 modem with asterisk Hi everyone, I just install Linux and asterisk on one of my pc. I want to run some basic functionality tests. Is it possible to use a v92 modem as a FXO or FXS card. If yes how do we configure and install the card? What are the commands? Thanks in advance for your help AC __ Lèche-vitrine ou lèche-écran ? magasinage.yahoo.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wellgate 3701
Hi everyone I'm trying to setup this Welltech Wellgate 3701 box. If I got to the proxy setup it seems to work but the Pstn incoming call always got a voice prompt from the Wellgate. Going to peer to peer mode seems to be better but I couldn't find any working configuration inside Asterisk. I do not really suffer from the registration problem because I doing all those trials with no password for the 3701 line configuration since I'm in a closed environment. Thanks for any help. Ml ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Buying some Polycom IP300s
Over the last few weeks/months I have been testing phones and ATAs from Grandstream (BT101, GXP2000, 286, 488), SNOM (190), Zyxel (Piece of Crap), Sipura (SPA-2000, SPA-841) and I personally feel that the Sipura SPA-841 is the best value, good quality phone that I have used. I haven't used the polycoms yet, but I plan to in the next few weeks. - Original Message - From: Dan Morin To: asterisk-users@lists.digium.com Sent: Sunday, April 03, 2005 10:22 AM Subject: [Asterisk-Users] Buying some Polycom IP300s Sorry =or the double post, I tried to paste and accidently sent the email I've been playing with Asterisk for a few =eeks now, and I've gotten everything to work well with softphones, so I'm ready to =ove on to normal VoIP phones. I've been looking around and reading =omments that people have had, and I was convinced that the Polycom IP300 was a great =hone for a good price. But, then I ran into this page, which has been =pdate in the last few days: http://w=w.voip-info.org/wiki-Polycom+SoundPoint+IP+500=DIV The page in the wiki used to say that the =erson would not recomed Polycom phones to anyone. So anyway, I just want to =ake sure that the IP300 is a good choice. I don't want to get cheap phones =hat aren't business quality, since I do play on using them for my business =fter testing. Also, is the IP500 worth the extra money? What can =t do that the IP300 can't. And finally, will the IP300 do ulaw encoding? Thanks in advance. ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-user ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI Dial Plan
Lee Lee wrote: Hi everyone Presently all our calls are channel to one provider and we would like to change that based on LCR. the following is what we have presently; # Dial the requested number, if we got something from the caller. if ($dialto != ) { $AGI-exec('SetAccount', $accountnum); if ($debug) { $AGI-exec('NoOp', \Dialing $dialto... \); } if ($dialto =~ /^416/) { $AGI-exec('Dial', Zap/g2/$dialto|30|C); } else { $AGI-exec('Dial', Zap/g1/$dialto|30|C) } } $AGI-hangup(); Cheers, Jean-Michel. -- Ykoz Un Max - La VoIP en pré-payé! Essayez gratuitement - 5 crédits offerts. --- http://ykoz.net/voip/max --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Buying some Polycom IP300s
My personal opinion is that the Polycom IP-300 is a slightly better phone than the Sipura, but I would be happy to be proved wrong on that. later, PaulH From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Monday, 4 April 2005 5:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Buying some Polycom IP300s Over the last few weeks/months I have been testing phones and ATAs from Grandstream (BT101, GXP2000, 286, 488), SNOM (190), Zyxel (Piece of Crap), Sipura (SPA-2000, SPA-841) and I personally feel that the Sipura SPA-841 is the best value, good quality phone that I have used. I haven't used the polycoms yet, but I plan to in the next few weeks. - Original Message - From: Dan Morin mailto:[EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, April 03, 2005 10:22 AM Subject: [Asterisk-Users] Buying some Polycom IP300s Sorry =or the double post, I tried to paste and accidently sent the email I've been playing with Asterisk for a few =eeks now, and I've gotten everything to work well with softphones, so I'm ready to =ove on to normal VoIP phones. I've been looking around and reading =omments that people have had, and I was convinced that the Polycom IP300 was a great =hone for a good price. But, then I ran into this page, which has been =pdate in the last few days: http://w=w.voip-info.org/wiki-Polycom+SoundPoint+IP+500 http://www.voip-info.org/wiki-Polycom+SoundPoint+IP+500 =DIV The page in the wiki used to say that the =erson would not recomed Polycom phones to anyone. So anyway, I just want to =ake sure that the IP300 is a good choice. I don't want to get cheap phones =hat aren't business quality, since I do play on using them for my business =fter testing. Also, is the IP500 worth the extra money? What can =t do that the IP300 can't. And finally, will the IP300 do ulaw encoding? Thanks in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-user CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk@Home Question
You would use the caller ID to route the call. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel Jafferali Sent: Sunday, April 03, 2005 10:17 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] Question I was wondering if there is a way to select the outbound trunk based on the extension that making the call. Set the context in the sip.conf file for that user to a context in extensions.conf that only has entries for dialing out through specific providers. Nabeel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Previous sip reload not yet done
Does anyone else have this problem? Is there a workaround? Yeah, I had this problem when I added a lot of SIP register statements and SIP peers. Changing the hostnames (FQDNs) to IP addresses solved the problem. It seem * was getting stuck waiting for DNS lookups. Nabeel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] at-320 phone configuration difficulty
Mishehu try 19750407 Also to get palmtool to work you need to play with the debug settings on the phone first. koltov Clive On 2 Apr 2005 at 0:30, I put the Who? in Mishehu wrote: Hi guys, I just got a Netweb 401 (AT-320) phone. It came with firmware 1.38 on it, and it has since been updated after failed attempts to configure, and now has 1.42 (IAX2) from centrality (P1688S). According to voip-info, atcom's docs, etc, there are two passwords for it - one is 1234, and the superuser password is supposed to be 12345678. Only 1234 works, and I get codec configuration, IP configuration, firmware/ringtone/dialplan update options. But nowhere do I find where to set information about my asterisk box I want this phone to connect to. I've tried using Palmtool 1.42, and anytime I try to query the phone's settings, I get Cannot connect to Palm1. The person who sold me sent no documentation or discs with it, and now on top of it, all the buttons such as Local Num and Local IP are all switched around. I am very unhappy, and have wasted 4 hours already trying to work on this. If anybody can assist, I'll be very grateful. -mishehu ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to send email from the dial plan?
I would like to get a notice by email, if we run out of gateways! exten = _9011Z.,410,Busy exten = _9011Z.,411,EMAIL = How to? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] error while compiling asterisk-1.0.7
gcc -shared -Xlinker -x -o cdr_odbc.so cdr_odbc.o -lodbc -L/usr/lib/pgsql gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DASTERISK_VERSION=\1.0.7\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -fPIC-c -o cdr_tds.o cdr_tds.c cdr_tds.c: In function `mssql_connect': cdr_tds.c:415: error: `TDSCONNECTINFO' undeclared (first use in this function) cdr_tds.c:415: error: (Each undeclared identifier is reported only once cdr_tds.c:415: error: for each function it appears in.) cdr_tds.c:415: error: `connection' undeclared (first use in this function) cdr_tds.c:460: warning: implicit declaration of function `tds_free_connect' cdr_tds.c: At top level: cdr_tds.c:71: warning: `connect_time' defined but not used make[1]: *** [cdr_tds.o] Error 1 make[1]: Leaving directory `/asterisk-1.0.7/cdr' make: *** [subdirs] Error 1 helo i am getting this error while compiling asterisk-1.0.7. any expert tell me what is this. regrads Kamran __ Yahoo! Messenger Show us what our next emoticon should look like. Join the fun. http://www.advision.webevents.yahoo.com/emoticontest ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Time sync on PRI
On Thu, 31 Mar 2005, Peter Svensson wrote: It would not be very hard to add both features to libpri. Libpri already has a function to decode and dump the time/date information. If I remember correctly the time/date IE should be added to the SETUP messages. I have been thinking about adding it, but have not had the time. It's already there, in bristuff patches. Please encourage Digium to add Junghanns' patches to the asterisk code :) -- Best Regards, Tobias Jönsson, Lund SE___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to send email from the dial plan?
Ronald Wiplinger wrote: I would like to get a notice by email, if we run out of gateways! exten = _9011Z.,410,Busy exten = _9011Z.,411,EMAIL = How to? -= Info about application 'System' =- [Synopsis]: Execute a system command [Description]: System(command): Executes a command by using system(). Returns -1 on failure to execute the specified command. If the command itself executes but is in error, and if there exists a priority n + 101, where 'n' is the priority of the current instance, then the channel will be setup to continue at that priority level. Otherwise, System returns 0. /O ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel group members - dial out on a availible port via trial and error?
Hi ya-all. Little question that has been bothering me somewhot. Say I have only 2 out going analog phone lines. Some1 in the office decides to call their a client... so the Dial command it using a group and it will start at the first Zap channel listed in the group. But now what if I disconnect the that line and he dials up again... then it tries the first Zap channel again - but why does it not time out or try the other Zap channel? Btw - I am using a TDM400P with the FXS ports on channel 3 and 4; *Asterisk* Urgent handler -- Executing Dial(Zap/1-1, Zap/g2/$EXTEN) in new stack Urgent handler Urgent handler -- Called g2/$EXTEN Urgent handler -- Zap/3-1 answered Zap/1-1 Urgent handler -- Attempting native bridge of Zap/1-1 and Zap/3-1 Urgent handler [And it hangs...] *extensions.conf* [outgoing] ;Dial 0 on the phone for external line ;SIP Phones need another was... they act like a cell phone exten = _0,1,Dial(Zap/g2/$EXTEN) ;exten = _0,2,NoOp(DIALSTATUS=${DIALSTATUS}) exten = _0,3,Goto(_0-${DIALSTATUS},1) ;Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten = _0-ANSWER,1,Goto(_0,102) exten = _0-.,1,Goto(_0,1) ;Try another line exten = _0,102,Congestion exten = _0,103,Hangup Ps - I cant get a log to see the DIALSTATUS. Also I would have expected that Asterisk would then just try the other group member but it does not Any help in this regard would be greatly apricaited. -- Kind Regards Etienne ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Previous sip reload not yet done
Nabeel Jafferali a écrit : Does anyone else have this problem? Is there a workaround? Yeah, I had this problem when I added a lot of SIP register statements and SIP peers. Changing the hostnames (FQDNs) to IP addresses solved the problem. It seem * was getting stuck waiting for DNS lookups. Thanks. We put everywhere it was accepted the IP address and will see. FYI, sipgate.de doesn't accept to register with IP address. CLI SIP reload command is now applied much faster as with FQDNs in sip.conf -- Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Q.931 to SIGTRAN interface
On 04/02/05 10:11 Mike Mueller said the following: I don't think an Asterisk box can generate enough calls to cause sockets related performance penalties. Five packets per phone call. What's the max call rate an Asterisk box can support? i think that would require an OS dependent answer. but generally, i'm more than interested in contributing resources towards an open sourced implementation of SS7 with asterisk. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.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 | +=+ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma VS. Digium
On 04/01/05 00:00 Matthew Boehm said the following: Steve Underwood wrote: And your EU bias is clearly demonstrated by this. I've never seen a BRI product outside he EU. :-) Come to Houston, TX. We were running a BRI for quite some time before upgrading to a T1. ahem, ISDN BRIs are fairly common here in asia too. but i guess that asia don't count now, does it ? :) -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.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 | +=+ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problems with call-forward from ccme to * on sip trunk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Uhmm ... maybe a connection plar from ccme to an * number (like 511 on my conf), then a simple forward from 511 to 601 on ccme? Something like: exten = _511,1,Dial(SIP/601,45) I need help ... :D Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFCUPWwMakHrsrHP9wRAuSVAKCMyKYIVSP8B+Tc0losELtmJovsEQCcDoOi gp1ZxZqe+G9hdAK6nEoqlaI= =D68e -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] how to configure groups using a sip phone
Hi Bacon Thanks for the quick response. Actually I want to confirm that whether it is possible to divide logical channels into group just like physiacl channels in zapata. Deepak Dhiman -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Monday, April 04, 2005 12:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] how to configure groups using a sip phone Can you be more specific? What are you trying to achieve with the creation of such groups? - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, April 04, 2005 3:50 PM Subject: [Asterisk-Users] how to configure groups using a sip phone hi friends ! i am facing a problem from one week and now required ur help urgently. Actually, i want to configure asterisk for two groups javgroup and linuxgroup. i also have constraint to use only sip phone (esatara ). now, please help me is it possible to configure astersik in that way or that kind of facility is given in zapata.conf. tell me in detail abt the configurations of the sip.conf and extensions.conf. thanks Deepak Dhiman ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Previous sip reload not yet done
administrator tootai wrote: Nabeel Jafferali a écrit : Does anyone else have this problem? Is there a workaround? Yeah, I had this problem when I added a lot of SIP register statements and SIP peers. Changing the hostnames (FQDNs) to IP addresses solved the problem. It seem * was getting stuck waiting for DNS lookups. Thanks. We put everywhere it was accepted the IP address and will see. FYI, sipgate.de doesn't accept to register with IP address. CLI SIP reload command is now applied much faster as with FQDNs in sip.conf Changing register= statements to IP addresses is a bad idea. SIP is domain name based and (as proved by sipgate) an IP address points to *one* host, whereas a SIP domain by using SRV records can point to many IP addresses and servers. There's a huge difference between sending a REGISTER to [EMAIL PROTECTED] and [EMAIL PROTECTED] See this as a short time fix. We need to make a better solution on the REGISTER parsing to prevent this from happening, it's clearly a bug. /O ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problems with call-forward from ccme to * on sip trunk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Apr 4, 2005, at 10:07 AM, Andrea Riela wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Uhmm ... maybe a connection plar from ccme to an * number (like 511 on my conf), then a simple forward from 511 to 601 on ccme? Something like: exten = _511,1,Dial(SIP/601,45) Ok, it works with this workaround ... But it's a workaround .. I hope some expert could help me to configure * correctly :) Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFCUP4IMakHrsrHP9wRAv5JAKCGor2S+v45KOs1g1mZ6iJiDWUgSQCgrupc z3UGbpMfaUbZf2ROxdxuW4U= =Rdk2 -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Previous sip reload not yet done
Olle E. Johansson a écrit : administrator tootai wrote: Nabeel Jafferali a écrit : Does anyone else have this problem? Is there a workaround? Yeah, I had this problem when I added a lot of SIP register statements and SIP peers. Changing the hostnames (FQDNs) to IP addresses solved the problem. It seem * was getting stuck waiting for DNS lookups. Thanks. We put everywhere it was accepted the IP address and will see. FYI, sipgate.de doesn't accept to register with IP address. CLI SIP reload command is now applied much faster as with FQDNs in sip.conf Changing register= statements to IP addresses is a bad idea. SIP is domain name based and (as proved by sipgate) an IP address points to *one* host, whereas a SIP domain by using SRV records can point to many IP addresses and servers. There's a huge difference between sending a REGISTER to [EMAIL PROTECTED] and [EMAIL PROTECTED] See this as a short time fix. We need to make a better solution on the REGISTER parsing to prevent this from happening, it's clearly a bug. Well noticed. Should I concider bugs #3850 and #3933 including this matter or should I open a new one? -- Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma VS. Digium
BRI's are in use in roughly 2/3 of the world with the US and I think China being the main exceptions. On Apr 4, 2005 9:37 AM, Dinesh Nair [EMAIL PROTECTED] wrote: On 04/01/05 00:00 Matthew Boehm said the following: Steve Underwood wrote: And your EU bias is clearly demonstrated by this. I've never seen a BRI product outside he EU. :-) Come to Houston, TX. We were running a BRI for quite some time before upgrading to a T1. ahem, ISDN BRIs are fairly common here in asia too. but i guess that asia don't count now, does it ? :) -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.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 | +=+ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michal Bielicki http://www.aefirion.org/ http://www.asterisk.com.pl/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Router with QoS recommendations
Hi, QoS is nice (and important) but only works within a FULLY controlled end to end link. Inside a BIG enterprise LAN, on leased lines its OK. Using end to end MPLS should also be ok Mind that some provider sell MPLS but it is not their own MPLS end to end. Going from one provider on MPLS to another on MPLS, you lose all the benefits. No control. Using the World Wide Wait (Internet) it will not help. A waste of money. My 2 cents. Shaoul Jacobson Senior VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk+Sipgate - just one step away..
Hello all, I have a working Asterisk setup, also a working sipgate.co.uk account (tested with a GrandStream ATA 486), but got stuck in forwarding calls from local users to sipgate. Very frustrating, since I feel there's just one silly error somewhere.. story follows: REGISTER both of the local user to * and of the * to sipgate.co.uk is successful but when dialing some random phone number in Linphone in the form sip:[EMAIL PROTECTED] (1.2.3.4 is the * box) I get -- Executing SetCallerID(SIP/user-733d, [EMAIL PROTECTED]) in new stack -- Executing Dial(SIP/user-733d, SIP/[EMAIL PROTECTED]|30|tr) in new stack Outgoing Call for is not a local user -- Called [EMAIL PROTECTED] Failed to authenticate on INVITE to ''[EMAIL PROTECTED] sip:[EMAIL PROTECTED];tag=as319c47a2' this I think is the problem - while the call is redirected, the correct number is dialed, Asterisk says it changed the callerid, but yy is the local username and 1.2.3.4 is the * address, shouldnt' it be ''[EMAIL PROTECTED] ? sip.conf: [general] register = :[EMAIL PROTECTED]/xx [sipgate] type=peer username= secret=pp host=sipgate.co.uk fromuser= fromdomain=sipgate.co.uk nat=no authuser= dtmfmode=info context=incomingsipgate context=default insecure=very canreinvite=yes disallow=all allow=ulaw allow=alaw extensions.conf: [general] static=yes writeprotect=yes [incomingsipgate] exten = h,1,Hangup exten = xxx,1,Dial(SIP/102,20,tr) [sipgate] exten = _9.,1,SetCallerID([EMAIL PROTECTED]) exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr) exten = _9.,3,Playback(invalid) exten = _9.,4,Hangup Any hints please? Thank you very much Razvan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Router with QoS recommendations
On 4 Apr 2005, at 09:25, Shaoul Jacobson - TELLINK wrote: Hi, QoS is nice (and important) but only works within a FULLY controlled end to end link. Inside a BIG enterprise LAN, on leased lines its OK. Using end to end MPLS should also be ok Mind that some provider sell MPLS but it is not their own MPLS end to end. Going from one provider on MPLS to another on MPLS, you lose all the benefits. No control. Using the World Wide Wait (Internet) it will not help. A waste of money. My 2 cents. I'm not sure I totally agree. It is also useful if you control the narrowest pipe. Take the example of several sub-offices joined to a head office PBX over 'public' ADSL lines. Let's say the company buys all the ADSL lines from the same provider. In such a set-up, the uplink side of the sub-office ADSL links are likely to be the main bandwidth limit. A well configured router there will slow outgoing email etc to preserve the quality of current VOIP sessions. Sure, the provider may have internal bandwidth constrictions, but they are unlikely to kick in before the 256k up channel of a typical ADSL. Oh, and, the web and the internet are not the same thing. Think like that and you'll forget mail. Which is a huge bandwidth consumer, and can stand being delayed by a second or two. Tim. http://www.westhawk.co.uk/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Planet VIP 450
Good day all Did someone get the planet VIP 450 working Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Detecting when a called mobile is not reachable?
I guess I should have added that this is based on the European, and specifically UK model, but I would have expected it to have been deemed best practice by most operators. On Apr 4, 2005 4:04 AM, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: Rod Bacon wrote: This is quite interesting. I tested calls to 2 mobiles that I knew were off, and not diverted to voicemail. 1 with Telstra, the other with vodafone (I'm in Australia). Via ISDN, both calls were shown as unanswered by asterisk. When the calls went to voicemail, the call was deemed to be answered. Via analogue circuits, the call is shown as answered, no matter what. That's what I would expect. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk+Sipgate - just one step away..
Razvan Cosma a écrit : Hello all, I have a working Asterisk setup, also a working sipgate.co.uk account (tested with a GrandStream ATA 486), but got stuck in forwarding calls from local users to sipgate. Very frustrating, since I feel there's just one silly error somewhere.. story follows: REGISTER both of the local user to * and of the * to sipgate.co.uk is successful but when dialing some random phone number in Linphone in the form sip:[EMAIL PROTECTED] (1.2.3.4 is the * box) I get -- Executing SetCallerID(SIP/user-733d, [EMAIL PROTECTED]) in new stack -- Executing Dial(SIP/user-733d, SIP/[EMAIL PROTECTED]|30|tr) in new stack Outgoing Call for is not a local user -- Called [EMAIL PROTECTED] Failed to authenticate on INVITE to ''[EMAIL PROTECTED] sip:[EMAIL PROTECTED];tag=as319c47a2' this I think is the problem - while the call is redirected, the correct number is dialed, Asterisk says it changed the callerid, but yy is the local username and 1.2.3.4 is the * address, shouldnt' it be ''[EMAIL PROTECTED] ? sip.conf: [general] register = :[EMAIL PROTECTED]/xx [sipgate] type=peer username= secret=pp host=sipgate.co.uk fromuser= fromdomain=sipgate.co.uk nat=no authuser= dtmfmode=info context=incomingsipgate context=default insecure=very canreinvite=yes disallow=all allow=ulaw allow=alaw extensions.conf: [general] static=yes writeprotect=yes [incomingsipgate] exten = h,1,Hangup exten = xxx,1,Dial(SIP/102,20,tr) [sipgate] exten = _9.,1,SetCallerID([EMAIL PROTECTED]) exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr) exten = _9.,3,Playback(invalid) exten = _9.,4,Hangup Any hints please? according to your sip.conf, should be [...] exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr) in extensions.conf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Manipulation based on SIP extension
Hello there, How do I configure any type of action based caller's extension and dialed number? For example if someone on extension 1777 calls extension 1777 this should be treated as accessing his voicemail box, so he won't need to call voicemail and entering mailbox number and password. I.N. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Difficulty in configuring Asterisk to ensure that Call Transfers (SIP phone) are properly recorded and billable
I am hoping someone in the * community has come across this problem before. Problem: Person SIP Phone A (SIPA) Person SIP Phone B (SIPB) SIP Phone C (SIPC PSTN Line) SIPA calls a billable phone number via SIPC exten = _123456/_1XX,1,SetAccount(${ACCOUNTCODE_COMPANYZ}) exten = _123456/_1XX,2,SetAMAFlags(billing) exten = _123456/_1XX,3,Macro(spa3k_pstn_out,${EXTEN}) Then SIPA, whom has generated charges decides to transfer the call the COMPANYY, and this is where the billing traceable problems begin; SIPA, transfers the call to SIPB, and the transaction information appears to be incorrect; a) SIPA is recorded as connecting with SIPB, and not SIPC. b) SIPC is then recorded as connecting with SIPB. Is there a way of ensuring that; a) SIPA transaction is recorded as SIPA called SIPC. b) SIPB transaction is recorded as SIPB called SIPC. A simular scenerio is also happening the other way round. i.e. outside dialling in via SIPC, SIPA answers, then transfers to SIPB. Also I need to ensure that the accountcode billing flags are correctly set when the call transfer has occurred, but I have not be sucessful as yet as I am lacking a bit of experience with the inner workings of configuration. If there is someone whom has experienced this problem before or something simular, I would be interested in knowing how you managed to resolve it. - Peter Info: -- Asterisk: v1.0.7 SIPA B: Polycom SoundPoint IP 300 SIPC:Sipura 3000 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Authenticating username
Nabeel, Could you expand on your comments, or provide a link / paste in a sample extensions.conf to show how this would be set up? David On Apr 4, 2005 12:57 AM, Nabeel Jafferali [EMAIL PROTECTED] wrote: Dial(SIP/904)calls whoever logged on as john. You could define a variable in extensions.conf. Nabeel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime voicemail
I tried to use ONE entry of my voicemail.conf to put into the database: [other] ;602=1357,Ronald Wiplinger 602,[EMAIL PROTECTED] INSERT INTO `voicemail_users` ( `uniqueid` , `customer_id` , `context` , `mailbox` , `password` , `fullname` , `email` , `pager` , `stamp` , `attach` , `saycid` , `hidefromdir` ) VALUES ('1', '602', 'other', '602', '3525', 'Ronald Wiplinger', '[EMAIL PROTECTED]', '', NOW( ) , 'no', 'yes', 'no') extconfig.conf includes: voicemail = mysql,astconf,voicemail_users *CLI reload -- Executing VoiceMail(SIP/601-a9a3, b602) in new stack Apr 4 17:48:34 WARNING[18977]: app_voicemail.c:2227 leave_voicemail: No entry in voicemail config file for '602' What do I miss? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music On Hold and ATA-186 w/Silence Supression
Hello, Alejandro! AG I have a problem with ATA-186 configured for silence supression Don't! I.N. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Difficulty in configuring Asterisk to ensure that Call Transfers (SIP phone) are properly recorded and billable
I am hoping someone in the * community has come across this problem before. Problem: Person SIP Phone A (SIPA) Person SIP Phone B (SIPB) SIP Phone C (SIPC PSTN Line) SIPA calls a billable phone number via SIPC exten = _123456/_1XX,1,SetAccount(${ACCOUNTCODE_COMPANYZ}) exten = _123456/_1XX,2,SetAMAFlags(billing) exten = _123456/_1XX,3,Macro(spa3k_pstn_out,${EXTEN}) Then SIPA, whom has generated charges decides to transfer the call the COMPANYY, and this is where the billing traceable problems begin; SIPA, transfers the call to SIPB, and the transaction information appears to be incorrect; a) SIPA is recorded as connecting with SIPB, and not SIPC. b) SIPC is then recorded as connecting with SIPB. Is there a way of ensuring that; a) SIPA transaction is recorded as SIPA called SIPC. b) SIPB transaction is recorded as SIPB called SIPC. A simular scenerio is also happening the other way round. i.e. outside dialling in via SIPC, SIPA answers, then transfers to SIPB. Also I need to ensure that the accountcode billing flags are correctly set when the call transfer has occurred, but I have not be sucessful as yet as I am lacking a bit of experience with the inner workings of configuration. If there is someone whom has experienced this problem before or something simular, I would be interested in knowing how you managed to resolve it. Info: -- Asterisk: v1.0.7 SIPA B: Polycom SoundPoint IP 300 SIPC:Sipura 3000 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk+Sipgate - just one step away..
On 04/04/2005 12:46 PM, administrator tootai wrote: according to your sip.conf, should be [...] exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr) in extensions.conf Ye :) Thank you very much! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Router with QoS recommendations
Hi, I'm not sure I totally agree. Good, we do agree on some :) I also agree with some of your remarks (no flame war) It is also useful if you control the narrowest pipe. I agree. But I disagree about the definition of the narrowest pipe. A well configured router there will slow outgoing email etc to preserve the quality of current VOIP sessions. agreed Let's say the company buys all the ADSL lines from the same provider. Buying all connection to the same provider is a wise decision. It does not give any guaranty but this can be discussed :) You are also a bigger customer. So you could negociate some QoS, sla, ... (read my thought after my sig) Most broadband (cable, xdsl) connection should provide enough bandwidth. If you use 70% or more of your bandwidth then I agree QoS will definitively help. (look during peaks for each up down link) Otherwise, not much. You share the bandwidth with other customers on your provider's backbone. And your ISP decides how to shape traffic. Some VoIP providers in the US are suing some ISP's because their VoIP traffic is degraded. The situation can be even worse with a cable connection as you share the bandwidth AT your end-point not at the backbone. Regards, Shaoul Jacobson Senior VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail [EMAIL PROTECTED] PS I have worked in close relations with some 'big' providers. They accept sla's, backup circuits even when they know they cannot provide. The customer is billed for this extra 'service' Extra billing is the only extra service the customer gets. Beside the false safety he things he got. If an accident happens, the isp pays for the lack of service. This is far cheaper than implementing the needed technology. I won't give names here, but this was the ways at some big international isp's, not a small local isp. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Supervised transfer problems
Hi all, when I try to transfer a call asterisk say me: -- Executing SetCallerID(SIP/20012-cb87, Gallina Daniele 20012) in new stack -- Executing Dial(SIP/20012-cb87, SIP/20013) in new stack -- Called 20013 -- SIP/20013-034d is ringing -- SIP/20013-034d answered SIP/20012-cb87 -- Attempting native bridge of SIP/20012-cb87 and SIP/20013-034d -- Started music on hold, class '3psystem', on SIP/20013-034d Apr 4 12:12:39 NOTICE[4710]: chan_sip.c:5161 get_refer_info: Supervised transfer requested, but unable to find callid '[EMAIL PROTECTED]' Apr 4 12:12:40 NOTICE[4710]: chan_sip.c:5161 get_refer_info: Supervised transfer requested, but unable to find callid '[EMAIL PROTECTED]' Apr 4 12:12:42 NOTICE[4710]: chan_sip.c:5161 get_refer_info: Supervised transfer requested, but unable to find callid '[EMAIL PROTECTED]' Why? Any ideas? Daniele -- Daniele Gallina 3P System S.r.l. - Software Developer Web: http://www.3psystem.net E-Mail: [EMAIL PROTECTED] Tel: 041.8626401 Scelta 2 Fax: 041.5161655 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Jitter buffer
Hi I am using CVS latest Is it correct there is no jitter buffer for SIP (RTP) Are there any plans for this? prob a stupid question: Is it required / do the endpoints handle this - if the src and destination are both SIP and there is no transcoding but asterisk is still in the media path? Thanks Jack __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do you do Line Hunting in Asterisk?
I have come accoross the fact that * can't handle if there is no dialtone So out of interist, can you do Line hunting in * in a sequencial manner and can you also do so in a random fasion? -- Kind Regards Etienne ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Time sync on PRI
On Mon, 4 Apr 2005, Tobias Jönsson wrote: On Thu, 31 Mar 2005, Peter Svensson wrote: It would not be very hard to add both features to libpri. Libpri already has a function to decode and dump the time/date information. If I remember correctly the time/date IE should be added to the SETUP messages. I have been thinking about adding it, but have not had the time. It's already there, in bristuff patches. Please encourage Digium to add Junghanns' patches to the asterisk code :) Since Junhhanns will not assign a transferable / resellable license to Digium and Digium will not accept any code that is not under such a license there is a bit of a stalemate. The Junghanns patch is not available for cvs head either. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zap - What is going on?
Ok - I was told that you set a group for Zap channels. So I tried to make use of my Zap channels so the 2 I am interisted in is channel 3 and channel 4. I make Channel 3 in use bu calling a line... then I try to call another line so expecting to have Zap channel 4 open and allowing me to make a call, but it just keeps on ringing... and then times out. Can anyone please shed some light on this for me? extensions.conf [outgoing] ;Dial 0 on the phone for external line ;SIP Phones need another way... they act like a cell phone exten = _0,1,Dial(Zap/g2/$EXTEN,20,tr) ;Try finding a line... exten = _0,2,Goto(_0-${DIALSTATUS},1) ;Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten = _0-ANSWER,1,Goto(_0,102) exten = _0-.,1,Goto(_0,1) ;Try another line exten = _0,102,Congestion exten = _0,103,Hangup Asterisk Console: -- Starting simple switch on 'Zap/1-1' -- Executing Dial(Zap/1-1, Zap/g2/$EXTEN|20|tr) in new stack -- Called g2/$EXTEN -- Nobody picked up in 2 ms -- Hungup 'Zap/4-1' ===That channel is free and has a seperate phone line connected to it. -- Executing Goto(Zap/1-1, _0-NOANSWER|1) in new stack -- Kind Regards Etienne ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@Home Question
I'd think about using a prefix for each trunk as a form of password. At home I have to dial 1 then the number to use one of my trunks, or 2 then the number for a different trunk. If you gave them a code of say 666 they would have to dial that then the number. If you had a code for your trunk they wouldn't be able to use your trunk unless they knew the code. Probably more elegant solutions but just a quick suggestion. tony I am using [EMAIL PROTECTED] 0.8 I was wondering if there is a way to select the outbound trunk based on the extension that making the call. Here is why I ask. Since I am already running my Asterisk server for my own use, I also wanted to let friends and family in on the action but I don't want to pay for their calls. So if I ask them to buy talk time from a termination provider and then setup a separate trunk for them, how do I make sure that only their calls use that outbound trunk? Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Manipulation based on SIP extension
- Original Message - From: Irakli Natsvlishvili [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 04, 2005 4:52 AM Subject: [Asterisk-Users] Manipulation based on SIP extension Hello there, How do I configure any type of action based caller's extension and dialed number? For example if someone on extension 1777 calls extension 1777 this should be treated as accessing his voicemail box, so he won't need to call voicemail and entering mailbox number and password. I.N. Try something like this exten = 1777,1,GotoIf($[${CALLERIDNUM} = 1777]?5:2) exten = 1777,2,Dial(SIP/177),15,rt exten = 1777,3,Voicemail(u${EXTEN}) exten = 1777,4,Hangup exten = 1777,5,VoicemailMain(s${EXTEN}) exten = 1777,6,Hangup Henry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: error while compiling asterisk-1.0.7
In article [EMAIL PROTECTED], Kamran Ahmad [EMAIL PROTECTED] wrote: gcc -shared -Xlinker -x -o cdr_odbc.so cdr_odbc.o -lodbc -L/usr/lib/pgsql gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DASTERISK_VERSION=\1.0.7\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -fPIC-c -o cdr_tds.o cdr_tds.c cdr_tds.c: In function `mssql_connect': cdr_tds.c:415: error: `TDSCONNECTINFO' undeclared (first use in this function) cdr_tds.c:415: error: (Each undeclared identifier is reported only once cdr_tds.c:415: error: for each function it appears in.) cdr_tds.c:415: error: `connection' undeclared (first use in this function) cdr_tds.c:460: warning: implicit declaration of function `tds_free_connect' cdr_tds.c: At top level: cdr_tds.c:71: warning: `connect_time' defined but not used make[1]: *** [cdr_tds.o] Error 1 make[1]: Leaving directory `/asterisk-1.0.7/cdr' make: *** [subdirs] Error 1 helo i am getting this error while compiling asterisk-1.0.7. any expert tell me what is this. Looks like it is trying to compile cdr_tds without the correct version of FreeTDS being installed. The makefile is finding /usr/include/tds.h or /usr/local/include/tds.h, which tells it to compile cdr_tds.c If you don't need to log CDRs to a MSSQL or Sybase server, the easiest solution if to comment out the two MODS+= lines under the heading FreeTDS stuff. If you do need it, then you will probably have to get a newer FreeTDS. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zap - What is going on?
For a start it should be ${EXTEN} You have to realize that ALL variables look like that. Dollar-open-curly-brackets-variablename-close-curly-brackets. So it didn't see your text as a variable and it tried to call the number $EXTEN on Zap/g2. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Etienne Pretorius Sent: 04 April 2005 13:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Zap - What is going on? Ok - I was told that you set a group for Zap channels. So I tried to make use of my Zap channels so the 2 I am interisted in is channel 3 and channel 4. I make Channel 3 in use bu calling a line... then I try to call another line so expecting to have Zap channel 4 open and allowing me to make a call, but it just keeps on ringing... and then times out. Can anyone please shed some light on this for me? extensions.conf [outgoing] ;Dial 0 on the phone for external line ;SIP Phones need another way... they act like a cell phone exten = _0,1,Dial(Zap/g2/$EXTEN,20,tr) ;Try finding a line... exten = _0,2,Goto(_0-${DIALSTATUS},1) ;Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten = _0-ANSWER,1,Goto(_0,102) exten = _0-.,1,Goto(_0,1) ;Try another line exten = _0,102,Congestion exten = _0,103,Hangup Asterisk Console: -- Starting simple switch on 'Zap/1-1' -- Executing Dial(Zap/1-1, Zap/g2/$EXTEN|20|tr) in new stack -- Called g2/$EXTEN -- Nobody picked up in 2 ms -- Hungup 'Zap/4-1' ===That channel is free and has a seperate phone line connected to it. -- Executing Goto(Zap/1-1, _0-NOANSWER|1) in new stack -- Kind Regards Etienne ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Previous sip reload not yet done
administrator tootai wrote: Olle E. Johansson a écrit : administrator tootai wrote: Nabeel Jafferali a écrit : Does anyone else have this problem? Is there a workaround? Yeah, I had this problem when I added a lot of SIP register statements and SIP peers. Changing the hostnames (FQDNs) to IP addresses solved the problem. It seem * was getting stuck waiting for DNS lookups. Thanks. We put everywhere it was accepted the IP address and will see. FYI, sipgate.de doesn't accept to register with IP address. CLI SIP reload command is now applied much faster as with FQDNs in sip.conf Changing register= statements to IP addresses is a bad idea. SIP is domain name based and (as proved by sipgate) an IP address points to *one* host, whereas a SIP domain by using SRV records can point to many IP addresses and servers. There's a huge difference between sending a REGISTER to [EMAIL PROTECTED] and [EMAIL PROTECTED] See this as a short time fix. We need to make a better solution on the REGISTER parsing to prevent this from happening, it's clearly a bug. Well noticed. Should I concider bugs #3850 and #3933 including this matter or should I open a new one? We had the same problem, on two different hardware platforms. 2 flavors of pentium 4/board combos grandstream and sipura (handset/ata) devices the only thing that has worked for us was to eliminate the registration process all together. This has been going on since last October that I am aware of which means it has been in every cvs since then. sip.conf host=device ip (not dynamic) qualify=yes ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sending faxes and call accounting
In the Asterisk system I am testing for implementation at a small luxury resort, there are four fax machines that the guests can use for sending and receiving faxes. Because they require confidentiality, we cannot use hylafax or other method than a stand alone fax. I would just connect these faxes to the PSTN lines directly but we would then have call accounting issues as the calls would not appear in the CDRs and with long distance costing from $1/min upwards, it could get costly quickly. How reliably can I do an analogue in/out connection with call accounting? I am using TDM400 cards or an Adtrans 600 channel bank. Thanks Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zap - What is going on?
Dope --- *sheepish grin*. Sorry. Thanks for the help. Kind Regards Etienne Technical Support Kingsley Technologies Rob Scott wrote: For a start it should be ${EXTEN} You have to realize that ALL variables look like that. Dollar-open-curly-brackets-variablename-close-curly-brackets. So it didn't see your text as a variable and it tried to call the number $EXTEN on Zap/g2. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Etienne Pretorius Sent: 04 April 2005 13:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Zap - What is going on? Ok - I was told that you set a group for Zap channels. So I tried to make use of my Zap channels so the 2 I am interisted in is channel 3 and channel 4. I make Channel 3 in use bu calling a line... then I try to call another line so expecting to have Zap channel 4 open and allowing me to make a call, but it just keeps on ringing... and then times out. Can anyone please shed some light on this for me? extensions.conf [outgoing] ;Dial 0 on the phone for external line ;SIP Phones need another way... they act like a cell phone exten = _0,1,Dial(Zap/g2/$EXTEN,20,tr) ;Try finding a line... exten = _0,2,Goto(_0-${DIALSTATUS},1) ;Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten = _0-ANSWER,1,Goto(_0,102) exten = _0-.,1,Goto(_0,1) ;Try another line exten = _0,102,Congestion exten = _0,103,Hangup Asterisk Console: -- Starting simple switch on 'Zap/1-1' -- Executing Dial(Zap/1-1, Zap/g2/$EXTEN|20|tr) in new stack -- Called g2/$EXTEN -- Nobody picked up in 2 ms -- Hungup 'Zap/4-1' ===That channel is free and has a seperate phone line connected to it. -- Executing Goto(Zap/1-1, _0-NOANSWER|1) in new stack ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Jitter buffer
I am using CVS latest Is it correct there is no jitter buffer for SIP (RTP) Are there any plans for this? prob a stupid question: Is it required / do the endpoints handle this - if the src and destination are both SIP and there is no transcoding but asterisk is still in the media path? My understanding is the new jitterbuffer code (in cvs-head) has been applied to iax connections, and the objective is to make it available for sip/rtp (and possibly other channel types) after things are cool in iax. A jitterbuffer is only required when the delivery of rtp packets is inconsistent (eg, jerky). Its my understanding that sip phones have at least some sort of jitterbuffer built into firmware. Don't know how effective they are for large variations in packet delivery though. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best way for nated sip peers thru a database
Hello list, Newbie questions Seems that nated sip peers/friends are not functional with RealTime because the database peers/users are not kept in memory. On the other side the dynamic config (MYSQL_FRIENDS) system does not support the nat option. Not sure but may be ast_data is the good way for that , or may be thru radius with PortaOne's Radius client ? Would like to know what's the best way to have nated peers in a database instead of flat files. Thanks in advance. Laurent -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.1 - Release Date: 01/04/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Previous sip reload not yet done
[...] See this as a short time fix. We need to make a better solution on the REGISTER parsing to prevent this from happening, it's clearly a bug. Well noticed. Should I concider bugs #3850 and #3933 including this matter or should I open a new one? We had the same problem, on two different hardware platforms. 2 flavors of pentium 4/board combos grandstream and sipura (handset/ata) devices the only thing that has worked for us was to eliminate the registration process all together. This has been going on since last October that I am aware of which means it has been in every cvs since then. Bug #3946 was open for this and Mantis. -- Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Supervised transfer problems
On Monday 04 April 2005 6:23 am, Daniele Gallina - 3P System S.r.l. wrote: Hi all, when I try to transfer a call asterisk say me: -- Executing SetCallerID(SIP/20012-cb87, Gallina Daniele 20012) in new stack -- Executing Dial(SIP/20012-cb87, SIP/20013) in new stack -- Called 20013 -- SIP/20013-034d is ringing -- SIP/20013-034d answered SIP/20012-cb87 -- Attempting native bridge of SIP/20012-cb87 and SIP/20013-034d -- Started music on hold, class '3psystem', on SIP/20013-034d Apr 4 12:12:39 NOTICE[4710]: chan_sip.c:5161 get_refer_info: Supervised transfer requested, but unable to find callid '[EMAIL PROTECTED]' snip I've been having the same 'problem' with my Polycom SoundPoint IP500 phone. When our receptionist hits transfer, ext, transfer - then I see a notice just as above. The odd thing is that the call ID is given as on the phone, not on the server. E.g. the IP in ''[EMAIL PROTECTED] is the IP of the Polycom phone, not the * box. Is there any way to fix this? Rewrite sip headers? Any ideas? -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: ASTCC question: Trunk LOCAL
Darren Wiebe wrote: That capability is not there yet. I would personally recommend using the 'Local' channel and routing your calls via the extensions.conf file. This is totally up to you but I find it gives me more flexibility. That would also make it easier to do something like you are looking to do with the setgroup and checkgroup commands. With the cards we have a field In Use. I would like to add this field to the TRUNK, so that I can jump to the next trunk instead, e.g., if I have several gateways available, but only a few ports at each gateway, than I need to jump to the next gateway. If I could add a field in Trunks, than I believe I could ask this field first, before I choose the trunk to dial, ... (not completely thought thru yet) Can you tell me more about it, please. It sounds interesting! In my extensions.conf I have the following lines: [default-outgoing] exten = _1NXXNXX,1,SetCIDNum(${CALLERIDNUM}|a) exten = _1NXXNXX,2,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN}) ... Then I have a trunk setup for the default-outgoing context. That way it uses the information above to route my calls. I would recommend looking the the wiki for the group commands as I've only used them a little bit. I tried this one, but it does not work!!! In Trunks I put in: 1-800-xxx Local Line-optimize I have a context [Line-optimize] but the real world says: * CLI -- AGI Script Executing Application: (DIAL) Options: (Local/011886228357765/Line-optimize|30|HL(59994:6:3)) -- Limit Data: -- timelimit=59994 -- play_warning=6 -- play_to_caller=yes -- play_to_callee=no -- warning_freq=3 -- start_sound=UNDEF -- warning_sound=timeleft -- end_sound=UNDEF Apr 4 20:26:55 NOTICE[1487]: chan_local.c:436 local_alloc: No such extension/context [EMAIL PROTECTED] creating local channel Apr 4 20:26:55 NOTICE[1487]: app_dial.c:936 dial_exec_full: Unable to create channel of type 'Local' (cause 0) == Everyone is busy/congested at this time (1:0/0/1) -- AGI Script astcc.agi completed, returning 0 Do you have any ideas??? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Realtime mysql problem?
Sorry for the delay, do you have any clue when realtime will get added to stable? I never did get this working but before I go too much further I'd like to run production on a stable version.. I'll try out SIP today and let you know, the reason I'm using IAX is because everything SIP we do is through SER. Not to mention since realtime doesn't support qualify= and NAT mode must be manually set, it's kind of pointless to use Asterisk for SIP. :-) Just for curiosity sake, have you tried any SIP RealTime stuff? Perhaps this is an IAX problem? I remember helping a guy a few weeks ago get his SIP RealTime working. This is the first IAX I've dealt with. And I have no IAX stuff to test with. -Original Message- From: Matthew Boehm [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 30, 2005 9:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime mysql problem? Matt Schulte wrote: How do you toggle the realtime cache? Check in the configs/iax.conf.sample file of a recent CVS download and it should be in there. If there were too many fields in the table, could you foresee this being a problem? No, because I have lots of extra company specific fields in my sip_users table that asterisk doesn't use at all and I've had no problems. ie iax users have peercontext and auth. Just for curiosity sake, have you tried any SIP RealTime stuff? Perhaps this is an IAX problem? I remember helping a guy a few weeks ago get his SIP RealTime working. This is the first IAX I've dealt with. And I have no IAX stuff to test with. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Router with QoS recommendations
tim panton [EMAIL PROTECTED] writes: On 4 Apr 2005, at 09:25, Shaoul Jacobson - TELLINK wrote: Hi, QoS is nice (and important) but only works within a FULLY controlled end to end link. Inside a BIG enterprise LAN, on leased lines its OK. Using end to end MPLS should also be ok Mind that some provider sell MPLS but it is not their own MPLS end to end. Going from one provider on MPLS to another on MPLS, you lose all the benefits. No control. Using the World Wide Wait (Internet) it will not help. A waste of money. My 2 cents. I'm not sure I totally agree. It is also useful if you control the narrowest pipe. Take the example of several sub-offices joined to a head office PBX over 'public' ADSL lines. Let's say the company buys all the ADSL lines from the same provider. In such a set-up, the uplink side of the sub-office ADSL links are likely to be the main bandwidth limit. A well configured router there will slow outgoing email etc to preserve the quality of current VOIP sessions. Sure, the provider may have internal bandwidth constrictions, but they are unlikely to kick in before the 256k up channel of a typical ADSL. Oh, and, the web and the internet are not the same thing. Think like that and you'll forget mail. Which is a huge bandwidth consumer, and can stand being delayed by a second or two. Tim. I agree, especially qos on upstream might be beneficial, and surely is in a cable modem setup. E.g. my modem has a 10 Mbit LAN interface, but uplink is limited to 256Kbit. So when I have many things going out, uplink will be much sooner saturated than the LAN link, and cable modem buffers run full leading to looong latencies and maybe even package loss. Putting a router before the modem shaping the upstream traffic solves that problem. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime mysql problem?
What is your problem with IAX in realtime? I have it working (finally). Wojtek - Original Message - From: Matt Schulte [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 04, 2005 9:01 AM Subject: RE: [Asterisk-Users] Realtime mysql problem? Sorry for the delay, do you have any clue when realtime will get added to stable? I never did get this working but before I go too much further I'd like to run production on a stable version.. I'll try out SIP today and let you know, the reason I'm using IAX is because everything SIP we do is through SER. Not to mention since realtime doesn't support qualify= and NAT mode must be manually set, it's kind of pointless to use Asterisk for SIP. :-) Just for curiosity sake, have you tried any SIP RealTime stuff? Perhaps this is an IAX problem? I remember helping a guy a few weeks ago get his SIP RealTime working. This is the first IAX I've dealt with. And I have no IAX stuff to test with. -Original Message- From: Matthew Boehm [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 30, 2005 9:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime mysql problem? Matt Schulte wrote: How do you toggle the realtime cache? Check in the configs/iax.conf.sample file of a recent CVS download and it should be in there. If there were too many fields in the table, could you foresee this being a problem? No, because I have lots of extra company specific fields in my sip_users table that asterisk doesn't use at all and I've had no problems. ie iax users have peercontext and auth. Just for curiosity sake, have you tried any SIP RealTime stuff? Perhaps this is an IAX problem? I remember helping a guy a few weeks ago get his SIP RealTime working. This is the first IAX I've dealt with. And I have no IAX stuff to test with. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk@Home Question
Message: 9 Date: Sun, 3 Apr 2005 23:52:39 -0500 From: * KAPIL * [EMAIL PROTECTED] Subject: [Asterisk-Users] [EMAIL PROTECTED] Question To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 Greetings! This is my first post to the list...and I'm kinda' new to Asterisk, so please be kindI did a fair amount of Googling but was not able to find an answer. I am using [EMAIL PROTECTED] 0.8 I was wondering if there is a way to select the outbound trunk based on the extension that making the call. Here is why I ask. Since I am already running my Asterisk server for my own use, I also wanted to let friends and family in on the action but I don't want to pay for their calls. So if I ask them to buy talk time from a termination provider and then setup a separate trunk for them, how do I make sure that only their calls use that outbound trunk? Any ideas? Set up a trunk for them Set up a route for them. Setup Outbound Routing Add Route Name your Route Dial Pattern 79|NXX 79|NXXNXX 79|1NXXNXX 79|011. Trunk Sequence Select there Trunk MAKE SURE they dial 79 for all there calls Jeff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk@Home Question
Greetings! This is my first post to the list...and I'm kinda' new to Asterisk, so please be kindI did a fair amount of Googling but was not able to find an answer. I am using [EMAIL PROTECTED] 0.8 I was wondering if there is a way to select the outbound trunk based on the extension that making the call. Here is why I ask. Since I am already running my Asterisk server for my own use, I also wanted to let friends and family in on the action but I don't want to pay for their calls. So if I ask them to buy talk time from a termination provider and then setup a separate trunk for them, how do I make sure that only their calls use that outbound trunk? Being rather new myself, but the first thing I thought about your problem was putting those extensions in a different context to the one where you define your trunk calls. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Realtime mysql problem?
Well, I made several posts. Basically realtime works fine on the system you register to, if you try to contact that peer from another Ast server (running realtime), it does a SELECT query and all finds the peer and continues to say Unable to contact peer as if the user doesn't exist. I even went as far as packet sniffing and noticed it doesn't ever go out on port 4569 or anything. Again, I've made several posts about this before for full details. :-) Thanks, Matt -Original Message- From: Wojciech Tryc [mailto:[EMAIL PROTECTED] Sent: Monday, April 04, 2005 8:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime mysql problem? What is your problem with IAX in realtime? I have it working (finally). Wojtek - Original Message - From: Matt Schulte [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 04, 2005 9:01 AM Subject: RE: [Asterisk-Users] Realtime mysql problem? Sorry for the delay, do you have any clue when realtime will get added to stable? I never did get this working but before I go too much further I'd like to run production on a stable version.. I'll try out SIP today and let you know, the reason I'm using IAX is because everything SIP we do is through SER. Not to mention since realtime doesn't support qualify= and NAT mode must be manually set, it's kind of pointless to use Asterisk for SIP. :-) Just for curiosity sake, have you tried any SIP RealTime stuff? Perhaps this is an IAX problem? I remember helping a guy a few weeks ago get his SIP RealTime working. This is the first IAX I've dealt with. And I have no IAX stuff to test with. -Original Message- From: Matthew Boehm [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 30, 2005 9:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime mysql problem? Matt Schulte wrote: How do you toggle the realtime cache? Check in the configs/iax.conf.sample file of a recent CVS download and it should be in there. If there were too many fields in the table, could you foresee this being a problem? No, because I have lots of extra company specific fields in my sip_users table that asterisk doesn't use at all and I've had no problems. ie iax users have peercontext and auth. Just for curiosity sake, have you tried any SIP RealTime stuff? Perhaps this is an IAX problem? I remember helping a guy a few weeks ago get his SIP RealTime working. This is the first IAX I've dealt with. And I have no IAX stuff to test with. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Router with QoS recommendations
On 03-Apr-2005, Tim Pushor wrote: I prefer PF's approach to security first, convenience second, and I *really* like the fact that PF has a real parser. As the requements get more complex, having everything in one file, and very readable and structured is a huge plus. Also, the integration with ALTQ is nice, especially for these types of applications. I agree with everything Tim wrote above, and I'll add that the biggest factor that influenced me in my move to OpenBSD for my firewall was that it was the only free unix I found that could do bidirectional filtering in bridged mode. As in, when you're in a bridged configuration you can filter in and out on an interface. Neither Linux nor FreeBSD could do this. It's certainly an edge case, but if you need that feature it's invaluable. I posted my asterisk altq experiments here: http://slacker.com/~nugget/asterisk4.php -- David McNett [EMAIL PROTECTED] http://slacker.com/~nugget/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI: received SETUP message for call that is not a new call, wicked!
Hi list, I'm getting the message... Apr 4 15:13:09 WARNING[1069]: chan_zap.c:7512 zt_pri_error: PRI: received SETUP message for call that is not a new call, wicked!!! This is running Asterisk 1.0.6-BRIstuffed-0.2.0-RC7k. These messages happen when someone calls from the Telco on a BRI line... but rather than asterisk simply immediately answering, they just hear ringing So really the new call IS a new call - but Asterisk things differently. Anyone met and/or solved this problem? This seems to be happening to 1 in 4 of all my calls??? - other calls are fine. -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] V92 modem with asterisk
Alexandre Charles said: Hi everyone, I just install Linux and asterisk on one of my pc. I want to run some basic functionality tests. Is it possible to use a v92 modem as a FXO or FXS card. If yes how do we configure and install the card? What are the commands? Thanks in advance for your help Most modems cannot be used as FXO's, since there are no drivers. The WC_FXO drivers works with SOME Intel chipsets. ebay: digium fxo, you'll find compatible modems for about $7 + SH. cheers, glenn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Router with QoS recommendations
I'm not sure about QoS, but I do run ATLQ on FreeBSD/PF. In a SOHO environment where there is likely to be DSL or cable, I find it very useful (on the upload side at least, which is usually a problem on asyncrhonous connections). I can max out my pipe and hear no effect of it on the phone. Shaoul Jacobson - TELLINK wrote: Hi, QoS is nice (and important) but only works within a FULLY controlled end to end link. Inside a BIG enterprise LAN, on leased lines its OK. Using end to end MPLS should also be ok Mind that some provider sell MPLS but it is not their own MPLS end to end. Going from one provider on MPLS to another on MPLS, you lose all the benefits. No control. Using the World Wide Wait (Internet) it will not help. A waste of money. My 2 cents. Shaoul Jacobson Senior VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Realtime Capabilities
Rod Bacon wrote: The term RTCache has never been mentioned in the WIKI or these forums. I assume that it's some sort of function to speed up realtime db access by keeping transactions in RAM and writing periodically? If so, I can understand why this would need to be flushed. RealTime Cache is a mechanism written to allow RealTime SIP/IAX peers/users to work with NAT and recieve MWI. When RTC is enabled, peer/user info is retreived from the database and stored in same list as sip.conf peers/users. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X-Lite to Zap, no Voice on other phone!
Hello, The problem is - and i was wandering if anyone knows the solution - is that When I dial from my windows machine, to an external phone line through Zap, then the receiving party does not hear my voice - but when the receiving party calls me back, then we have voice on both sides. What makes the more currious is that internal numbers work fine, both sides have voice. -- Kind Regards Etienne ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime mysql problem?
do you have any clue when realtime will get added to stable? It won't. Not to mention since realtime doesn't support qualify= and NAT mode must be manually set, Have you been using RTC? (RealTime Cache) It fixes the NAT/MWI problem. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXy audio troubles (only on INCOMING calls)
Hello All! I just got my IAXy.. Configured it.. Got it Up and Running Calls OUT have no problems (that means from IAXy - Asterisk - ZAP/SIPclient/IAXclient) Calls IN do have problems (that means from ZAP/SIPclient/IAXclient - Asterisk - IAXy) On those incoming calls on my IAXy I hear the other party on my IAXy, But this other party can't hear me (the audio that's beeing sent from the IAXy to asterisk can't be heard) Does anyone have any idea what I can do about this? Is this a common problem? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best way for nated sip peers thru a database
Laurent FOULONNEAU wrote: Hello list, Newbie questions Seems that nated sip peers/friends are not functional with RealTime because the database peers/users are not kept in memory. *sigh* I'm quoting this wiki page: http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Sip quoteNOTE: As of CVS-HEAD 3/16/05, if you enable RealTime caching in your sip.conf, Voicemail MWI works and so does 'sip show peers'./quote This also enables NAT because using the cache, peers/users are kept in memory. Would like to know what's the best way to have nated peers in a database instead of flat files. You could always do this for your sip.conf: http://www.voip-info.org/wiki-Asterisk+RealTime+Static -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Browser based configuration of Asterisk
Hi, I have a Linux Fedora 3 Asterisk only box (2 FXO 2 FXS ports) with no GUI or WEB server running. I can get to it remotely using Putty but I want to add the capability to at least do Dial Plan configuration via a browser. Do any of the GUI based configurations support such a setup. I really do not want to install a GUI on the Asterisk box but installing Apache would be okay... I googled ( :list.digium.com) around the mailing lists and found several boxes that do this (Mediatrix and Epygi) but I do want to have to pay for a box when I already have an old server converted. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Router with QoS recommendations
At 15:36 04/04/2005, you wrote: On 03-Apr-2005, Tim Pushor wrote: I prefer PF's approach to security first, convenience second, and I *really* like the fact that PF has a real parser. As the requements get more complex, having everything in one file, and very readable and structured is a huge plus. Also, the integration with ALTQ is nice, especially for these types of applications. I agree with everything Tim wrote above, and I'll add that the biggest factor that influenced me in my move to OpenBSD for my firewall was that it was the only free unix I found that could do bidirectional filtering in bridged mode. As in, when you're in a bridged configuration you can filter in and out on an interface. Neither Linux nor FreeBSD could do this. It's certainly an edge case, but if you need that feature it's invaluable. I'm using ALTQ since FreeBSD 4.6 and it's also exist ALTQ+PF that's near the same as OpenBSD version. And i confirm that's shapping with ALTQ work great ! Even with 32 Kbps. You can easely shape around 1000 rules and have a full Fast Ethernet port on a dual PIII (FreeBSD ALTQ port without PF) ALTQ have many shape algo, maybe the only one with such diversity. You have some CD distribution with ALTQ enable. I posted my asterisk altq experiments here: http://slacker.com/~nugget/asterisk4.php -- David McNett [EMAIL PROTECTED] http://slacker.com/~nugget/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Arnaud Pignard ([EMAIL PROTECTED]) Frontier Online - Opérateur Internet ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Absolute Timeout
Hi, I dial a number with following setting: exten = _X.,1,Absolutetimeout(20)exten = _X.,2,dial(SIP/[EMAIL PROTECTED]|L(30))exten = T,1,BackGround(tt-weasels)exten = T,2,Hangup() I find Absolute time out is not working , is it normal? kaiser ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Detecting when a called mobile is not reachable?
David John Walsh wrote: I guess I should have added that this is based on the European, and specifically UK model, but I would have expected it to have been deemed best practice by most operators. On Apr 4, 2005 4:04 AM, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: Rod Bacon wrote: This is quite interesting. I tested calls to 2 mobiles that I knew were off, and not diverted to voicemail. 1 with Telstra, the other with vodafone (I'm in Australia). Via ISDN, both calls were shown as unanswered by asterisk. When the calls went to voicemail, the call was deemed to be answered. Via analogue circuits, the call is shown as answered, no matter what. That's what I would expect. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thaks for all your replies, adding the *r* seems to help. Ian. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP phones to Asterisk using MAC address instead of IP address
Hi, I know this can be done but I guess I am not understanding the few notes I have seen on this - can SIP phones be tied to Asterisk using a PC mac address instead of their IP address (obviously I am using DHCP). If someone could please point to the right Wiki or How to I would greatly appreciate it. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problems with call-forward from ccme to * on sip trunk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 To help to find my mistake, I've two debugs: 1) isdn -- connection plar to 5600 on * -- 601 on cme -- vm call-forward to 5601 on * ext.num 123456789 calls my ISDN number, on ccme there's a connection plar to internal 5600 (on asterisk), that dials automatically to 601 on ccme. The call-forward noan (no answer) to 5601 on * works great debug: http://www.nesys.it/sipwork.txt 2) isdn -- connection plar to 601 on cme -- vm call-forward to 5601 * ext.num 123456789 calls my ISDN number, on ccme there's a connection plar directly to internal 601. The call-forward noan to 5601 doesn't work correctly (the call goes to *, but the connection tears down) debug: http://www.nesys.it/sipdnwork.txt I think that's a debug quite interesting for all sip people. Any advice will be appreciated Thanks for your support Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFCUUl+MakHrsrHP9wRAjaWAJ9TX38RK7w8UxYSC52w8mKAU3vTjACgzSNl lcsr7AsP5qC4MZrvEdcAldc= =XAdb -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime mysql problem?
can you send me a dump from SQL for this account? I have it working both ways, W - Original Message - From: Matt Schulte [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 04, 2005 9:34 AM Subject: RE: [Asterisk-Users] Realtime mysql problem? Well, I made several posts. Basically realtime works fine on the system you register to, if you try to contact that peer from another Ast server (running realtime), it does a SELECT query and all finds the peer and continues to say Unable to contact peer as if the user doesn't exist. I even went as far as packet sniffing and noticed it doesn't ever go out on port 4569 or anything. Again, I've made several posts about this before for full details. :-) Thanks, Matt -Original Message- From: Wojciech Tryc [mailto:[EMAIL PROTECTED] Sent: Monday, April 04, 2005 8:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime mysql problem? What is your problem with IAX in realtime? I have it working (finally). Wojtek - Original Message - From: Matt Schulte [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 04, 2005 9:01 AM Subject: RE: [Asterisk-Users] Realtime mysql problem? Sorry for the delay, do you have any clue when realtime will get added to stable? I never did get this working but before I go too much further I'd like to run production on a stable version.. I'll try out SIP today and let you know, the reason I'm using IAX is because everything SIP we do is through SER. Not to mention since realtime doesn't support qualify= and NAT mode must be manually set, it's kind of pointless to use Asterisk for SIP. :-) Just for curiosity sake, have you tried any SIP RealTime stuff? Perhaps this is an IAX problem? I remember helping a guy a few weeks ago get his SIP RealTime working. This is the first IAX I've dealt with. And I have no IAX stuff to test with. -Original Message- From: Matthew Boehm [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 30, 2005 9:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime mysql problem? Matt Schulte wrote: How do you toggle the realtime cache? Check in the configs/iax.conf.sample file of a recent CVS download and it should be in there. If there were too many fields in the table, could you foresee this being a problem? No, because I have lots of extra company specific fields in my sip_users table that asterisk doesn't use at all and I've had no problems. ie iax users have peercontext and auth. Just for curiosity sake, have you tried any SIP RealTime stuff? Perhaps this is an IAX problem? I remember helping a guy a few weeks ago get his SIP RealTime working. This is the first IAX I've dealt with. And I have no IAX stuff to test with. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sending faxes and call accounting
Chris Mason (Lists) said: In the Asterisk system I am testing for implementation at a small luxury resort, there are four fax machines that the guests can use for sending and receiving faxes. Because they require confidentiality, we cannot use hylafax or other method than a stand alone fax. I don't understand you're confidentiality arguement. If asterisk is switching the call, it /can/ save a copy of the transmission. None the less, you should be able to switch a fax call just like a voice call. cheers, glenn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distributed services such as voicemail using Asterisk
Hi, Is it possible to distribute services used by Asterisk onto several boxes - similar to Pingtel (Pingtel is not an option for me since I need to tie analog phones into the system). The main service I want to distribute is the voice mail. I know that Mysql (have not tried PostGreSQL yet) can be used for voice mail and configuration files but can the MySQL server be on another server??? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Realtime - extensions configuration help
Hi, The wiki http://www.voip-info.org/wiki-Asterisk+RealTime+Extensions shows a very trivial sample: INSERT INTO `extensions_table` VALUES (1, 'mycontext', '_574555', 1, 'Wait', '2'); but how would you 'translate' an old definition as : exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr) I already found out that the commas need to be replaced by '|'. (exten = ... Dial(SIP/1007,20,tr) becomes ..., 'Dial', '1007|20|tr' ) It is mentioned only for the 'goto' in the wiki. Maybe is it worth to broaden up the sample. Regards, Shaoul Jacobson Senior VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP phones to Asterisk using MAC address insteadof IP address
Hi, I know this can be done but I guess I am not understanding the few notes I have seen on this - can SIP phones be tied to Asterisk using a PC mac address instead of their IP address (obviously I am using DHCP). If someone could please point to the right Wiki or How to I would greatly appreciate it. I would do this by using IP reservations on the DHCP server. Most DHCP servers will allow you to set a reservation of a paricular IP address to a particular MAC address. You may not be able to use this if you have more phones than available IP addresses of course. I couldn't see anything in http://www.voip-info.org/wiki-Asterisk+config+sip.conf that would help your cause directly. Giles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: X100P interrupt load
On Mon, 2005-04-04 at 12:29 +0800, Dinesh Nair wrote: On 03/23/05 04:15 Jesse Guardiani said the following: This should be has some issues. I do not consider the FreeBSD zaptel support to be production quality in any way. I experienced reproducible system hangs (mostly after an asterisk restart), interrupt issues (audio skips and SSH pauses during typing), and general instability. This was with an up-to-date FreeBSD 5.3-SECURITY and the latest zaptel at asterisk from ports (1.0.6 for asterisk, and a significantly lower version for zaptel, I think). I do not recommend anyone run FreeBSD + Asterisk at this time. perhaps a post detailing how these hangs happenned and any CLI output before these hangs would help in /eliminating/ this. :) I doubt it. The zaptel driver for FreeBSD isn't up-to-date with the Linux version, so I doubt Zaptel support on FreeBSD will ever be quite as reliable as Linux. But if you're curious: the hangs could be forced by restarting the asterisk server. Sometimes it would survive one restart then crash on the second. Sometimes it would crash for no reason at all. i'm running asterisk on freebsd 4.x /with/ digium TDM cards without any problems. any problems i faced were usually tied down the the digium hardware itself, instead of asterisk or freebsd. note that noload = pbx_wilcalu needs to exist in modules.conf, as detailed in the asterisk on freebsd wiki. not a hardware guy, so I don't know much about interrupts. Just that 1000 interrupts/sec is fairly high. :) those are the interrupts which the digium cards generate, and are used for timing. it's not specifically a freebsd issue. -- Jesse Guardiani, Systems Administrator WingNET Internet Services, P.O. Box 2605 // Cleveland, TN 37320-2605 423-559-LINK (v) 423-559-5145 (f) http://www.wingnet.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] configuring md5 authentication
Hello, How does md5 authentication works? I have created a user on my sip.conf like this: [dov]type=friendhost=dynamicusername=dovauth=md5; echo -n "dov:myhost.com.br:dov" | md5summd5secret=a72d3b44ea28fc6515d922b21970b83c ;secret=dov Where myhost is the real that I normally use on my SIP phone when I don't use md5 authentication. The echo line is the command I used to convert my user:realm:pwd into md5. In my X-Ten phone I just enter my username "dov" and password "dov" as plain text. It doesn't log in as I thought it should... Is there any extra setting that I have to define? Thank you Dov ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Realtime mysql problem?
do you have any clue when realtime will get added to stable? It won't. why not? Not to mention since realtime doesn't support qualify= and NAT mode must be manually set, Have you been using RTC? (RealTime Cache) It fixes the NAT/MWI problem. I haven't tried this yet because of my other issue, is RTC on by default? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] configuring md5 authentication
Where myhost is the real that I normally use on my SIP phone when I don't use md5 authentication. The echo line is the command I used to convert my user:realm:pwd into md5. In my X-Ten phone I just enter my username dov and password dov as plain text. It doesn't log in as I thought it should... Is there any extra setting that I have to define? Did you use the same realm, that was specified in the sip.conf? The default is asterisk. -- Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP signature.asc Description: Digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie - want to use asterisk as an internal PBX
Hallo. At the begining I would like to use asterisk as a VoIP server for some internal extensions inside one building without connection to external world. I planning to use kphone as soft phones. I tried to use configureation description that is described in http://asterisk.net.au/tutorial/1/ I'm running RH7.3, compiled and installed asterisk successfuly, compiled kphone. I set up all scripts according to the link above. /usr/sbin/asterisk -vvvgc : seems to be starging OK. When I run kphone I am able to login: trace from asterisk console: == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/enum.conf': Found == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 1 - 2 Asterisk Ready. *CLI -- Registered SIP 'kphone' at 10.1.3.154 port 5062 expires 180 Problems start now. According to extensions.conf: ; echo test, to make sure your phone works. exten = 600,1,Playback(demo-echotest) ; Let them know what's going on exten = 600,2,Echo ; Do the echo test exten = 600,3,Playback(demo-echodone) ; Let them know it's over exten = 600,4,Goto(s,6) ; Start over I GUESS that when I dial 600, I should be able to hear echo when I'm talking, but unfortunatelly I cannot even dial the number. I tried many ways (10.1.3.154 - is my asterisk pbx): 600 [EMAIL PROTECTED] sip:[EMAIL PROTECTED] all the dials above produce asterisk log message: Apr 4 16:54:25 NOTICE[27916]: pbx.c:1329 pbx_extension_helper: Cannot find extension context 'voipmenu' and kphone says: call failed: not found My sound card works fine. I do not know what can I do more. Have You got any ideas. Greetings Jeste pracodawc? Szukasz pracownika? Zamie ogoszenie w Praca.wp.pl! Internet to skuteczne narzdzie rekrutacyjne! http://klik.wp.pl/?adr=http%3A%2F%2Fpraca.wp.pl%2Fzamiesc.htmlsid=345 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Realtime - extensions configuration help
Shaoul Jacobson - TELLINK wrote: Hi, The wiki http://www.voip-info.org/wiki-Asterisk+RealTime+Extensions shows a very trivial sample: INSERT INTO `extensions_table` VALUES (1, 'mycontext', '_574555', 1, 'Wait', '2'); but how would you 'translate' an old definition as : exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr) I already found out that the commas need to be replaced by '|'. (exten = ... Dial(SIP/1007,20,tr) becomes ..., 'Dial', '1007|20|tr' ) It is mentioned only for the 'goto' in the wiki. Maybe is it worth to broaden up the sample. You just answered your own question in the same post. so..why did you even post this question if you answered it 2 lines later? -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Just a test
Title: Just a test Just testing our new subscription. Ping J Rick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Does the agent queue app support Aftercall and AUX agent status?
In most call centers I have worked in, the agents had the ability to change their status from "auto ready" or "available" into an AUX of After call state, Aftercall basically works like wrap time, in that the agent would not receive another call in the queue until their status was manually changed back to auto ready by a specific key combination on the dial pad. Aux worked in that the agent could change their state from auto ready into an AUX state where the press a code that indicates what type of AUX state they are in, an example would be Aux-Break, or Aux-Supervisor feedback (for tracking of time, etc.) Example, I am an agent, I receive a call, while on the call, I dial a special key code, and then when the call disconnects, instead of going right back to the queue and receiving another call, I go into an After Call state, where I can write notes, and log the call, etc. Then I would dial another sequence to put me back into a ready state. An example of Aux, I am an agent, I am in a queue, but not on a call, my supervisor calls my extension and says they need to discuss one of my previous calls with me. I dial a code, and my state is changed from auto ready into an Aux state, where I then dial an additional digit to indicate why I am in Aux, example: Feedback, Break, etc. I then get back to my desk, and dial a new code to place me back into a ready state. I know that with Asterisk, you can program a wrap time option to allow the CSR X number of seconds or minutes of wrap time before receiving another call, but I am looking for the above functionality over and above the simple implementation of wrap time. I do not want to just have the agents log in, and out when they don't want a call, but instead use the functionality I described above as a time keeping system for payroll, reporting, and agent tracking purposes. I sent an email, with a more ambiguous subject line about this subject, and received no response, so I am hoping with a better subject line, someone may open the email. In that previous email I mentioned that on the digium homepage's FAQ it listed some call center terminology that detailed the above mentioned functionality, but I can not find and documentation on it, so I am hoping it exists, but has not been documented yet, and that someone out there has used it, or knows if it truly does exist, or if I am out of luck. The link to the FAQ section: http://www.digium.com/index.php?menu=faq#General_7 If I am asking the wrong list, if someone knows a better place to ask this question, please let me know. Thanks, Steve MannNetwork AdministratorFineLine Solutions ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Snom and Multiple calls
Okay, after talking with Sven today, it turns out my problem description is wrong (I was combining to cases, one of which does work in the current firmware): - Multiple incoming calls (works already) - Incoming call while dialing (or waiting for answer of) outgoing call (doesn't) -- Joshua P. Dady smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@Home Question
yip, i think that is the best approach. --Dalon On Apr 4, 2005 6:33 AM, Giles Coochey [EMAIL PROTECTED] wrote: Greetings! This is my first post to the list...and I'm kinda' new to Asterisk, so please be kindI did a fair amount of Googling but was not able to find an answer. I am using [EMAIL PROTECTED] 0.8 I was wondering if there is a way to select the outbound trunk based on the extension that making the call. Here is why I ask. Since I am already running my Asterisk server for my own use, I also wanted to let friends and family in on the action but I don't want to pay for their calls. So if I ask them to buy talk time from a termination provider and then setup a separate trunk for them, how do I make sure that only their calls use that outbound trunk? Being rather new myself, but the first thing I thought about your problem was putting those extensions in a different context to the one where you define your trunk calls. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem registering 'SJPhone'?
Hi, Has anyone had problems registering an SJPone software phone. I get lots of junk mail so I have some filters running in Thunderbird and I have not seen my registration acknowledge come through. Do they (www.sjlabs.com) use some other domain for registration?? Any one else had this problem?? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk-1.0.7 Build - Serious issues
On Mar 31, 2005 1:24 PM, Dana Olson [EMAIL PROTECTED] wrote: On Thu, 31 Mar 2005 10:04:34 -0500, Kanuri, Seshu (Company IT) [EMAIL PROTECTED] wrote: Folks! I want to let everyone know that I have been trying to migrate from 1.0.6 to 1.0.7 last few days and I have come across serious issues in the build 1.0.7. What I found are listed below. I would recommend everyone to hold off any upgrade till the next build. 1)Voicemail - No Audio. Asterisk is not able to stream the voice to the Uas. 0-9 Digit files seem to be missing and Asterisk does not try to say extension numbers for the called user. My guess is all these .gsm files are corrupt and hence you don't hear anything. 2)Music on hold - .MP3 files in the ../mohmp3 and other folders are corrupt. When we tried to play these files using a media player, all we hear is gibberish. 3)DTMF is screwed up. Whatever worked in 1.06 does not work now when we configure this for RFC2833. Has anyone upgraded to 1.0.7 from 1.0.6 and had these issues and been able to find a fix? Seshu Just to add to this, I've been having some issues with audio with 1.0.7 as well. I haven't yet downgraded back to 1.0.6 to see if it solves it, but basically, I'm hearing some artifacts, and these didn't occur in the last 3 or 4 builds. I'm using the same phone and same config as I had been on 1.0.6. Other people that I call say that my phone sounds like crap now too. Also, when dialing over a Zap channel, the audio seems to sorta stutter now, at the very first second or two of a call. This didn't happen prior to 1.0.7. When I get some time that the server is not in use, I will downgrade and try to confirm that it is definitely an issue with the new build, but so far it seems that way. -- Dana I rolled back to Asterisk 1.0.6 this morning and things seem back to normal as far as voice quality goes. -- Dana ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Router with QoS recommendations
Any FreeBSD/OpenBSD solutions we should add to the list at the bottom of this page? http://www.voip-info.org/tiki-index.php?page=VOIP+Routers Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Arnaud PIGNARD To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, April 04, 2005 3:57 AM Subject: Re: [Asterisk-Users] Router with QoS recommendations At 15:36 04/04/2005, you wrote:On 03-Apr-2005, Tim Pushor wrote: I prefer PF's approach to security first, convenience second, and I *really* like the fact that PF has a real parser. As the requements get more complex, having everything in one file, and very readable and structured is a huge plus. Also, the integration with ALTQ is nice, especially for these types of applications.I agree with everything Tim wrote above, and I'll add that the biggestfactor that influenced me in my move to OpenBSD for my firewall was thatit was the only free unix I found that could do bidirectional filteringin bridged mode. As in, when you're in a bridged configuration you canfilter in and out on an interface. Neither Linux nor FreeBSD could dothis. It's certainly an edge case, but if you need that feature it'sinvaluable.I'm using ALTQ since FreeBSD 4.6 and it's also exist ALTQ+PF that's near the same as OpenBSD version.And i confirm that's shapping with ALTQ work great ! Even with 32 Kbps.You can easely shape around 1000 rules and have a full Fast Ethernet port on a dual PIII (FreeBSD ALTQ port without PF)ALTQ have many shape algo, maybe the only one with such diversity.You have some CD distribution with ALTQ enable.I posted my asterisk altq experiments here: http://slacker.com/~nugget/asterisk4.php--David McNett [EMAIL PROTECTED]http://slacker.com/~nugget/___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Arnaud Pignard ([EMAIL PROTECTED])Frontier Online - Opérateur Internet___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma VS. Digium
On Mar 31, 2005 1:44 PM, Dana Olson [EMAIL PROTECTED] wrote: On Thu, 31 Mar 2005 11:37:19 -0600, Rich Adamson [EMAIL PROTECTED] wrote: My understanding is that to an extent when we buy Sangoma we're putting the dagger to Digium. If anything puts the dagger to Digium it'll be their own inability to engineer reliable hardware. I appreciate what Digium has done for Asterisk, but reliability expectations for phone equipment are extremely high. I sympathize with people who need hardware that doesn't need to be restarted once a week just to do its job properly. If Digium can't deliver on those reliability expectations, and do it soon, people are going to switch to companies that can. And you know what? I don't blame them. The Digium boards need to be restarted once a week? Please clarify this. I was dead set on getting in a Sangoma A104 for a production Asterisk box, but then I read this thread and felt that it didn't matter so much what I would order... And so I was deciding to stick with Digium. And then I read your scary comment. I've currently got a Digium board filled with 3 T1s, but it hasn't been under heavy use right yet, due to my attention being pulled from * and put onto SER+AudioCodes devices for other applications, and I haven't had to restart yet. Is this going to change? What's the deal? Please clarify your statement for me, as I need reliability as well. I'll jump in here (but I'm not the original poster). The once a week thing relates to the digium TDM card (fxo and/or fxs modules). I don't believe the T1 cards are an issue that requires driver reloads. Alright, that helps clarify it a bit, but then again, I have been running Asterisk at home with a TDM card for a couple months and haven't had to restart it for a long time. Is it a requirement or just simply a recomendation? I shouldn't have said anything. My incoming pots line stopped responding this weekend. I found out when I got an email from someone telling me that it just keeps ringing and ringing. This never happened before... Strange thing is though, today, my IAX number wasn't responding. It's like I'm silently losing my registration or something. Totally unrelated, I know. I'm gonna go back to 1.0.6 because I ran it for a good while with no problems. I hope this solves it. It fixed the issues I had with 1.0.7 at work. -- Dana ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime mysql problem?
Matt Schulte wrote: do you have any clue when realtime will get added to stable? It won't. why not? Now, this has been answered many, many, many times...in fact..I believe Olle answered this in his Welcome to Asterisk post he sent out over the weekend. To summarize: The stable branch is for bug fixes only. New features will never be added to stable. If you want new AND stable wait for 1.2. I haven't tried this yet because of my other issue, is RTC on by default? No. You might want to check your configs/sip.conf.sample for the correct settings. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users