RE: [Asterisk-Users] Cannot open chan_zap:
Any why would that make it work with cvs-head but not cvs-stable? By the way, I no_load the module so I can load it manually later and see the console output. Either way, it still kicks out the error and crashes, or just kicks out the error if I no_load it first... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower Sent: Tuesday, April 12, 2005 12:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cannot open chan_zap: Tim Connolly wrote: Well crapola... cvs-head works with Digium's te110xp, but not cvs stable. Looks like there's a huge difference: Stable=-rw--- 1 root root 248572 Jun 9 2004 chan_zap.c Head =-rw--- 1 root root 326585 Apr 6 14:17 chan_zap.c I run a te110p with 1.0.x CVS stable all the time. You have a problem with your modules.conf and forgot to put the .so on the load = line. ___ 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] (no subject)
Hi, I have just bought another TDM400P card from Digium directly, purchased last Thursday, received it today: Span 1: WCTDM/0 Wildcard TDM400P REV E/F Board 1 1 WCTDM/0/0 FXOKS (In use) 2 WCTDM/0/1 FXOKS (In use) 3 WCTDM/0/2 FXSKS (In use) 4 WCTDM/0/3 FXSKS (In use) So it seems that this is the same problem. I also find it strange that this card also has some intermittend issues with audio quality. We here are currently testing multiple different PBX solutions from Nortel BCM through Avaya and cisco to asterisk. Asterisk deffinitly has potential, though when cards have sound issues, that aint great. I hope Digium will send some sort of firmware upgrade procedure, if even possible. S. On Mon, 11 Apr 2005, Robert Webb wrote: Good morning all.. I was following a discussion on this list about the TDM400P revisions. It is my understanding that the current revision that one should have is the Rev. H and not the E/F. I have not yet been able to verify the rev stamped on the board, but zaptel is reporting that I have the Rev. E/F. I just bought this card in January direct from Digium and was wondering if I got the wrong Rev. somehow?? I have been having some intermittent problems but only thought it was my setup. ___ 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] Has anyone got Asterisk working behind a NAT connection to users within a NAT
I was wondering if anyone has managed to get a working solution with asterisk behind a NAT connecting to external sip users behind another NAT. I have been using iax for the asterisk box without any issues including internal and external connections as well as connecting multiple asterisk boxes together with shared dial plans. However, and I know this is a running issues, I cannot get external sip users behind a NAT to be able to successfully connect to asterisk when it's behind a NAT as well. The external sip user correctly registers with asterisk and I can dial their phone and they can dial into to any phone internal to the network. They can even hear my conversation when they use an external stun server (in this case stun.xten.net) but they cannot hear me. When debugging I realise that the ip addresses are not pointing at the right internal server and hence the traffic never gets through. I have been looking at using ser but after compiling and testing it I never found a solution owing to it's complexity and the possibility that I would need to get ser to answer calls on port 5060 and potentially redirect them to another port (say 5061) for the sip phones. I have done port forwarding at both ends dealing with the usual ports of 5060, 4569 and 5036 as well as opening up the rtp ports for the voice traffic on 10,000 to 20,000. Is there a way without asterisk being on an external ip? Any help would be useful especially if someone has managed to get a working system. Thanks or your help. Regards Fats -- Fats Neutron [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] Problem with * transfer
Hi, I make a call to my mobile, now I would like to transfer the call to another extension from my mobile, I try with #1 (which is configured in features.conf as unattended transfer), and pbxtransfer is played back to me, but when I try to enter an extension I just get an error. What am I doing wrong? This is the entries from the CLI: -- Playing 'pbx-transfer' (language 'da') -- Unable to find extension '2' in context '' -- Playing 'beeperr' (language 'da') It seems that it's trying to fin extension 2 in context - how do I set the context? I am dialing through IAX2 and this is the entry in iax.conf: [_MTk4MzA4NA_pP3CqDqJvz] language=da type=friend host=129.142.224.250 secret=consealed context=default canreinvite=no notransfer=no trunk=no disallow=all allow=ulaw allow=alaw allow=ilbc allow=gsm regards - Thorben ___ 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: Problems trying to compile H323 from CVS-STABLE
Tony Mountifield wrote: Yesterday I wrote: I'm trying to compile channels/h323 and chan_h323 from CVS-STABLE, on Fedora Core 3. [... snip ...] Well I gave up with chan_h323, which is a pity, because it should be the solution that is better integrated with Asterisk. I would still like to hear from anybody that has any ideas (please see my original post). Instead, I downloaded asterisk-oh323-0.6.5 from InAccessNetworks, along with Janus-patch4 of PWlib (1.6.6.3) and OpenH323 (1.13.5.3). Following the instructions exactly, installation went smoothly, and worked first time. When testing the ability of dual 3GHz Xeons to handle many simultaneous OH323 calls (G.711 so no heavy transcoding), I discovered that chan_oh323 is EXTREMELY profligate with file descriptors! Each open oh323 channel uses 21 fds, yes TWENTY-ONE! In order to handle upwards of 120 simultaneous calls I needed to increase the per-process file descriptor limit from the default of 1024, using the technique described at: http://www.xenoclast.org/doc/benchmark/HTTP-benchmarking-HOWTO/node7.html I then added ulimit -n 8192 to /usr/sbin/safe_asterisk. It seems to be working ok now, but I'd still like to get chan_h323 working sometime, as I have a feeling it will be much less hungry for file descriptors! Comments, anyone? We are working on pushing this number down. Be patient! Michael. ___ 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 quit abnormally
Hi all, I have a VoIP PBX box with asterisk and one x100p card. I setup some sip users in sip.conf. The asterisk will quit aperiodically, sometime it will work for several days before quit, but I find its quit time is almost in 18:00 to 19:00. I can not find any clue from log file. The asterisk log is attached. Could anyone help me? Thanks a lot. --- Best regards, Qiao Yuansong mailto: [EMAIL PROTECTED] messages Description: Binary data ___ 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] About Audio Latency from PSTN to SIP
Hi all, I built a VoIP PBX box with asterisk and one x100p card. Every thing is ok except there is a short audio latency from PSTN to SIP and no delay in the reverse direction. At the beginning of a call, the latency is not very long, but it becomes more and more obvious along with time. If the call keep 10 minutes, the delay will be about half or one second. Anyone knows the reason, and any suggestion? Thanks a lot. --- Best regards, Qiao Yuansong mailto: [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] Best FXO Voip Gateway for Asterisk
Quintum are good Selon Chad Brown [EMAIL PROTECTED]: There are many analogue gateways to choose from: http://www.voip-info.org/wiki-VoIP+Gateways Does anyone have experience with several that could point me in the right direction? I need 5-8 ports. At some point I see us going digital but I'm not sure when TCO will make sense. Advice based on real world experience would be much appreciated. Thanks, Chad ___ 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] Re: Problems trying to compile H323 from CVS-STABLE
In article [EMAIL PROTECTED], Michael Manousos [EMAIL PROTECTED] wrote: Tony Mountifield wrote: When testing the ability of dual 3GHz Xeons to handle many simultaneous OH323 calls (G.711 so no heavy transcoding), I discovered that chan_oh323 is EXTREMELY profligate with file descriptors! Each open oh323 channel uses 21 fds, yes TWENTY-ONE! We are working on pushing this number down. Be patient! I'll look forward to it - thanks! It would be nice if any such improvements are made available in a version compatible with Stable as well as with Head. 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
[Asterisk-Users] Asterisk on HP DL380 G4 - chan_zap.so problems
Hi there! I compiled asterisk on a HP DL380 G4 with Suse Linux Enterprise Server 9 (gcc 3.3.3). It compiled without any errors. I also had no problems with installing my digium hardware (WC TE110P). But when I try to start asterisk, I get the following error messages: The error messages Apr 12 10:22:37 WARNING[8756]: chan_iax2.c:4796 timing_read: Unable to acknowledge zap timer .. == Parsing '/etc/asterisk/zapata.conf': Found Apr 12 10:22:37 WARNING[8756]: chan_zap.c:924 zt_open: Unable to specify channel 1: Inappropriate ioctl for device Apr 12 10:22:37 ERROR[8756]: chan_zap.c:6460 mkintf: Unable to open channel 1: Inappropriate ioctl for device here = 0, tmp-channel = 1, channel = 1 Apr 12 10:22:37 ERROR[8756]: chan_zap.c:10247 setup_zap: Unable to register channel '1-15' Apr 12 10:22:37 WARNING[8756]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Apr 12 10:22:37 WARNING[8756]: loader.c:440 load_modules: Loading module chan_zap.so failed! Can anybody help me on this? I would really appreciate that :) Luke PS: Sorry for my English Loaded modules: Module Size Used by snd_pcm_oss 65704 0 snd_pcm 112900 1 snd_pcm_oss snd_page_alloc 16264 1 snd_pcm snd_timer 32260 1 snd_pcm snd_mixer_oss 24448 1 snd_pcm_oss snd 71012 4 snd_pcm_oss,snd_pcm,snd_timer,snd_mixer_oss soundcore 13536 1 snd edd 13720 0 joydev 14528 0 sg 41760 0 st 45212 0 sr_mod 21028 0 ide_cd 42628 0 cdrom 43036 2 sr_mod,ide_cd nvram 13448 0 wcte11xp28448 0 zaptel 188420 1 wcte11xp hw_random 9620 0 ehci_hcd33668 0 uhci_hcd35728 0 thermal 16648 0 processor 21568 1 thermal fan 8196 0 button 10384 0 evdev 13952 0 battery 12804 0 ipv6275580 17 ac 8964 0 raw 44064 0 usbcore 116700 4 ehci_hcd,uhci_hcd tg3 80516 0 isdn145612 0 slhc11392 1 isdn subfs 12160 1 dm_mod 59904 0 ext3123688 1 jbd 75172 1 ext3 cciss 47332 3 sd_mod 25088 0 scsi_mod120132 5 sg,st,sr_mod,cciss,sd_mod ztcfg -vv: Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: Individual Clear channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: Individual Clear channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: Individual Clear channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: Individual Clear channel (Default) (Slaves: 12) Channel 13: Individual Clear channel (Default) (Slaves: 13) Channel 14: Individual Clear channel (Default) (Slaves: 14) Channel 15: Individual Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Individual Clear channel (Default) (Slaves: 17) Channel 18: Individual Clear channel (Default) (Slaves: 18) Channel 19: Individual Clear channel (Default) (Slaves: 19) Channel 20: Individual Clear channel (Default) (Slaves: 20) Channel 21: Individual Clear channel (Default) (Slaves: 21) Channel 22: Individual Clear channel (Default) (Slaves: 22) Channel 23: Individual Clear channel (Default) (Slaves: 23) Channel 24: Individual Clear channel (Default) (Slaves: 24) Channel 25: Individual Clear channel (Default) (Slaves: 25) Channel 26: Individual Clear channel (Default) (Slaves: 26) Channel 27: Individual Clear channel (Default) (Slaves: 27) Channel 28: Individual Clear channel (Default) (Slaves: 28) Channel 29: Individual Clear channel (Default) (Slaves: 29) Channel 30: Individual Clear channel (Default) (Slaves: 30) Channel 31: Individual Clear channel (Default) (Slaves: 31) 31 channels configured. My /etc/zaptel.conf loadzone=nl defaultzone=nl span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 My /etc/asterisk/Zapata.conf [channels] language=de context=default switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no
[Asterisk-Users] Voicemail quota
Hello, Is there a way to put a voicemail quota to a SIP user? I mean a quota on the user's mailbox instead of a particular message of the user like the 'maxmessage' does currently. Quata can be total file size of message or total minutes of messages of a mailbox. Thanks Foong ___ 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] Supervisor monitor / barge in - automatically on call setup?
I'm aware of the legal issues surrounding my request, but any help technically would be greatly apreciated On site we have a fully staffed hospital and fire service (its a temporary event for a childrens charity) and an onsite 911 number. If a user dials the number, they goto the emergency crew, and the use of monitor helps to record the call - thats the easy bit I'm in the UK, and its an offence not to pass a 999 (our 911) call out to a 999 centre but with the sheer numbers involved, we have a few choices, only one of which is suitable. If a user inadvertantly dials 999 I would like to pass it to the true 999 and at the same time dial either a special phone, or all the phones in the emergency centre. Upon the centre answering it, it silently monitors the call between the user and the 999 centre. If for whatever reason the centre needs to barge in they can, prehaps even silencing the origninal user. We have a 2 min response time to anywhere on site, the offical user services have about 22, but we know and expect that in a moments panic someone will dial the number automatically Any assistance as to how this can be performed will be greatly apreciated. ___ 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] Supervisor monitor / barge in - automatically on call setup?
David John Walsh wrote: I'm aware of the legal issues surrounding my request, but any help technically would be greatly apreciated On site we have a fully staffed hospital and fire service (its a temporary event for a childrens charity) and an onsite 911 number. If a user dials the number, they goto the emergency crew, and the use of monitor helps to record the call - thats the easy bit Send the call into a conference call, I'm in the UK, and its an offence not to pass a 999 (our 911) call out to a 999 centre but with the sheer numbers involved, we have a few choices, only one of which is suitable. If a user inadvertantly dials 999 I would like to pass it to the true 999 and at the same time dial either a special phone, or all the phones in the emergency centre. Upon the centre answering it, it silently monitors the call between the user and the 999 centre. If for whatever reason the centre needs to barge in they can, prehaps even silencing the origninal user. We have a 2 min response time to anywhere on site, the offical user services have about 22, but we know and expect that in a moments panic someone will dial the number automatically Any assistance as to how this can be performed will be greatly apreciated. ___ 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 -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ 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] Petition for IAX firmware
http://www.petitiononline.com/mod_perl/signed.cgi?IAXPhone Sorry for the 170 or so who have already signed. This list supposedly has 10,000 or more subscribers. 170 isn't very impressive. Please sign! ___ 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 quit abnormally
Qiao Yuansong wrote: Hi all, I have a VoIP PBX box with asterisk and one x100p card. I setup some sip users in sip.conf. The asterisk will quit aperiodically, sometime it will work for several days before quit, but I find its quit time is almost in 18:00 to 19:00. That is the time to go home, right? Was before Windows installed on that hardware? (of course it is just a joke) bye Ronald I can not find any clue from log file. The asterisk log is attached. Could anyone help me? Thanks a lot. --- Best regards, Qiao Yuansong mailto: [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 -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ 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: Problems trying to compile H323 from CVS-STABLE
Tony Mountifield wrote: In article [EMAIL PROTECTED], Michael Manousos [EMAIL PROTECTED] wrote: Tony Mountifield wrote: When testing the ability of dual 3GHz Xeons to handle many simultaneous OH323 calls (G.711 so no heavy transcoding), I discovered that chan_oh323 is EXTREMELY profligate with file descriptors! Each open oh323 channel uses 21 fds, yes TWENTY-ONE! We are working on pushing this number down. Be patient! I'll look forward to it - thanks! It would be nice if any such improvements are made available in a version compatible with Stable as well as with Head. We maintain versions compatible with both the stable and HEAD branches of Asterisk. Michael. ___ 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] Has anyone got Asterisk working behind a NAT connection to users within a NAT
However, and I know this is a running issues, I cannot get external sip users behind a NAT to be able to successfully connect to asterisk when it's behind a NAT as well. I have done port forwarding at both ends dealing with the usual ports of 5060, 4569 and 5036 as well as opening up the rtp ports for the voice traffic on 10,000 to 20,000. Is there a way without asterisk being on an external ip? Are you using nat=yes in sip.conf entries and giving the externip and localnet parameters? ___ 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 Addons compile errors
HI: I have compiled and installed Asterisk 1.0.7 without any problems.I have also installed mysql and DBD::mysql successfuly / When I tried to make asterisk-addons, it showed me the problem like these: [EMAIL PROTECTED] asterisk-addons]# make install ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` app_addon_sql_mysql.c:162:64: macro AST_LIST_REMOVE requires 4 arguments, but only 3 given make -C format_mp3 all make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o common.o common.c gcc -pipe -fPIC -Wall -WstricUntitled 1t-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o dct64_i386.o dct64_i386.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o decode_ntom.o decode_ntom.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o layer3.o layer3.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o tabinit.o tabinit.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o interface.o interface.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o format_mp3.o format_mp3.c format_mp3.c: In function `load_module': format_mp3.c:335: warning: passing arg 5 of `ast_format_register' from incompatible pointer type gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -shared -Xlinker -x -o format_mp3.so common.o dct64_i386.o decode_ntom.o layer3.o tabinit.o interface.o format_mp3.o make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3' cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_saycountpl.o app_saycountpl.c cc -shared -Xlinker -x -o app_saycountpl.so app_saycountpl.o cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c: In function `my_load_module': cdr_addon_mysql.c:269: warning: assignment makes pointer from integer without a cast cc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o -lmysqlclient -lz -L/usr/lib/mysql cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:162:64: macro AST_LIST_REMOVE requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:162: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:162: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:162: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 I am not a Linux Expert. What can I do for make addons? Thank to all, and sorry for my poor english. _ Do You Yahoo!? http://cn.rd.yahoo.com/mail_cn/tag/1g/*http://cn.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] H.323 Question
Hello list, I have a question about Asterisk and H323. Wich H323 channel driver is the best for Asterisk? Asterisk-oh323 or OH323. Im asking this question because I have big problem running my asterisk with asterisk-oh323. all is well installed but when there are some calls, my asterisk stop running. Right nowm Im using asterisk-v1.0.2 LSE RPM distro with all the modules ( asterisk-addons, asterisk-oh323, asterisk-zaptel, asterisk-libpri). All these modules are RPMs but I still have the same problem. Ive first used version 1.0.RC2 of asterisk and corresponding modules. Can some body help me with this issue. Regards. 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] TE110P - NT-Mode ?
Hello, I still try to connect a TE110P card to a TMS2 card in a Siemens HiPath 3750. The TMS2 card can be used to connect to an NT (Amtsanschluss) or to connect to another S2M-Line (PRI). When connecting to another PRI, I can select between CorNet (proprietary), ECMA-QSIG and ISO-QSIG. It seems that Asterisk supports none of these protocols. When connecting to an NT, I select Euro-Amt PP, but then the TMS2-Card expects a NT (Network Termination) at the other side. Is there any way I can switch the TE110P card to NT-Mode ? Regards, 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
RE: [Asterisk-Users] Changing DTMF mode depending on codec chosen
No offense taken. In fact, it sounds like you have 'spotted' an error or potential error in the way I have configured this. I would appreciate any and all comments/suggestions you may have on how I could configure asterisk to change dtmfmode depending on the codec being used. Thanks, - Andre -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Luki Sent: Monday, April 11, 2005 11:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Changing DTMF mode depending on codec chosen It seems that sometimes broadvoice honors my g.729 request Be careful with this. I tried setting G726-32 as a prefered codec and SOME calls would accept it (depending on call destination) but usually the caller did NOT hear me, although I could hear the caller just fine. So there's truth to it: BV officially only supports G711 so use it to avoid surprises... and end up blaming Broadvoice :). Their service is actually quite good and reliable, but unfortunately proper configuration seems to challenge many Asterisk users -- this is a general observation, nothing personal. No wonder Broadvoice doesn't officially support Asterisk... --Luki ___ 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] Changing DTMF mode depending on codec chosen
Hi Rich, Thanks for writing back to me. Yep, just like you, I too am looking for a lower bandwidth codec for my outbound. And, yes, broadvoice only officially supports G.711. That being said, is there even a way to do this scenario in asterisk? Thanks, - Andre -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson Sent: Monday, April 11, 2005 10:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Changing DTMF mode depending on codec chosen I'm not quite sure if this can be done, but.. I use BROADVOICE as my outbound primary. I have both g.729 and ulaw as my outbound preference with BROADVOICE. It seems that sometimes broadvoice honors my g.729 request, and that is the codec chosen for the outbound call via broadvoice.. Other times, I get ulaw. The problem is, when I'm using g.729 I get 'inband is not supported on this codec'.. And when I'm using ulaw, rfc2833 doesn't seem to work.. Am I doing something wrong, or is it true that the dtmfmode has to change depending on which codec is being used. And, if that is the case, how can I tell asterisk to change the dtmfmode for the call depending on which codec is being used for the call? Unless broadvoice changed something, they only officially support g711 (not g729). So, config your system to only allow ulaw and dtmf=inband for broadvoice. The g711 requirement is why I gave up and discontinued their service since I have limited dsl bandwidth on my end. ___ 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] Line Noise HELP!
rusty*CLI show version
Asterisk CVS-HEAD-03/26/05-17:05:44 built by [EMAIL PROTECTED] on a
i686 running Linux
rusty*CLI
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rich
Adamson
Sent: Monday, April 11, 2005 6:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Line Noise HELP!
And, what asterisk version are you running?
Ooops, sorry folks.. A correction..
I don't have digium X100 cards, I have Digit Networks X100P clone cards..
Don't know if it
matters, but wanted to get the facts straight :-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andre Normandin
Sent: Monday, April 11, 2005 5:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Line Noise HELP!
Hi,
I'm having very similiar problems.. However, I'm running a development
version, and it
happens on both SIP phones, and on Analog
phones connected via Sipura SPA-2000's (I have 2 different SPA2000's,
and 4 analog lines..
Seems to happen on all of them as well)..
The problem seems to be EXACTLY as described.
THe call seems fine at first, then within minutes the call degrades to
the point that
neither end can hear each other.. First, the volume
seems to lower, and then static, breaking up, etc..
I have both DIGIUM X100 cards for my pots lines (3 of them), and
BROADVOICE for outgoing
calls. It seems to happen no matter if I'm
on an analog line (I.E. someone called me), or if it was me that
initiated the call
(BROADVOICE outbound).
I do have a 'remote' SIPURA SPA2000 located at a friends house in a
different state -- he
is an extension on my pbx so he can call me, and
he can call his friends locally (He just moved away) via my POTS or
BROADVOICE line.. He
experiences the same problems as I described
above, unless he calls me directly at my 'internal' extension, or I
call him at his
'internal' extension.. I.E. If it doesn't touch POTS or
BROADVOICE, the problem doesn't seem to occur..??
The other interesting thing that has happened of recent development is
that some people
are complaining that they are hearing the
'electronic beep' sound as if the call is being recorded, but I am not
recording the call.
This has occured with my friend as well as incoming
and outgoing POTS/BROADVOICE calls.
If anyone has an idea, I'd love to hear it.. The problem is driving me
(and others who
talk to me) crazy!!!
- Andre
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Damian Funnell
Sent: Monday, April 11, 2005 3:08 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Line Noise HELP!
Hi mate,
Interesting - you're using a different version of Asterisk than I
am, but your problem
sounds identical. We thought it was the SIP
phones that we were using as well, but then it started occurring
on the analogue
phones as well.
Post again when you've tried a new phone, will you? Let us know
if the problem goes
away.
Cheers,
Damian.
Paul wrote:
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Damian,
pbx*CLI show version
Asterisk CVS-HEAD-03/23/05-00:44:07 built by [EMAIL PROTECTED] on
a
i586 running Linux
There is my version info. Someone on the list has suggested
that its my SIPura
phone. It could very well be the phone, but it
just seems unlikely that the conversation would be perfectly
clear for some time
before the static starts. I tried
[Asterisk-Users] Problem with fxo
Hi, i am trying to use my fxo card for analog calls .. fxo card seems to be ok, working properly but when trying to call outside ( from a sip phone ot pstn ) i get the following error on asterisk . Apr 12 11:59:24 DEBUG[4231]: chan_sip.c:4633 build_route: build_route: Contact hop: Drugo sip:[EMAIL PROTECTED]:5060 -- Executing Dial(SIP/69-562c, Zap/1/651559526|5) in new stack Apr 12 11:59:24 DEBUG[4231]: chan_zap.c:1645 zt_call: Dialing '651559526' Apr 12 11:59:24 DEBUG[4231]: chan_zap.c:1706 zt_call: Deferring dialing... -- Called 1/651559526 Apr 12 11:59:25 DEBUG[4231]: chan_zap.c:3770 __zt_exception: Exception on 15, channel 1 Apr 12 11:59:25 DEBUG[4231]: chan_zap.c:3082 zt_handle_event: Got event Hook Transition Complete(12) on channel 1 (index 0) Apr 12 11:59:27 DEBUG[4231]: chan_zap.c:3770 __zt_exception: Exception on 15, channel 1 Apr 12 11:59:27 DEBUG[4231]: chan_zap.c:3082 zt_handle_event: Got event Dial Complete(9) on channel 1 (index 0) Apr 12 11:59:27 DEBUG[4231]: chan_zap.c:1224 zt_enable_ec: Enabled echo cancellation on channel 1 Apr 12 11:59:27 DEBUG[4231]: channel.c:1363 ast_read: Dropping duplicate answer! any clue? got no info about exception 15 :/ 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-users
Re: [Asterisk-Users] (no subject)
The only firmware upgrade procedure is for you to call digium support. Hi, I have just bought another TDM400P card from Digium directly, purchased last Thursday, received it today: Span 1: WCTDM/0 Wildcard TDM400P REV E/F Board 1 1 WCTDM/0/0 FXOKS (In use) 2 WCTDM/0/1 FXOKS (In use) 3 WCTDM/0/2 FXSKS (In use) 4 WCTDM/0/3 FXSKS (In use) So it seems that this is the same problem. I also find it strange that this card also has some intermittend issues with audio quality. We here are currently testing multiple different PBX solutions from Nortel BCM through Avaya and cisco to asterisk. Asterisk deffinitly has potential, though when cards have sound issues, that aint great. I hope Digium will send some sort of firmware upgrade procedure, if even possible. S. On Mon, 11 Apr 2005, Robert Webb wrote: Good morning all.. I was following a discussion on this list about the TDM400P revisions. It is my understanding that the current revision that one should have is the Rev. H and not the E/F. I have not yet been able to verify the rev stamped on the board, but zaptel is reporting that I have the Rev. E/F. I just bought this card in January direct from Digium and was wondering if I got the wrong Rev. somehow?? I have been having some intermittent problems but only thought it was my setup. ___ 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 ---End of Original Message- ___ 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] Version 0.80 of IPS released
Version 0.80 - 12. April 2005. * Swedish language added - thanks Daniel Nylander * Bug fixes Download for FREE: http://ipswitchboard.thorben.dk Would you like to help translate IPS into your language? Please click the link below for details. I will add your language as soon as I receive it. http://ipswitchboard.thorben.dk/index.php?option=com_simpleboardItemid=42f unc=showcatcatid=5 IPSwitchBoard is an Operators Panel for the Asterisk PBX. IPSwitchBoard is a FREE Windows.NET application which gives you: * Unattended/attended transfers. * Park calls and retrieve/forward them again. * Organize all your SIP and IAX extensions (automatically retrieved from Asterisk). * Monitor all extensions. * Monitor all queues. * Monitor Agents. * Monitor Parked Calls. * Dynamically log extensions in and out of queues. * Integration with CRM software on the web. * Drop any active call. * Import/Export extensions to/from Asterisk Server DB. * Set Do Not Disturb on Extensions and give a reason. * Speed Dialling. ___ 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] Petition for IAX firmware
Now it would be even more interesting to see if Cisco or maybe Siemens/Polycom would bring out a firmware for IAX, now that would be a revolution.. :) On Tue, 12 Apr 2005, Wilson Pickett wrote: http://www.petitiononline.com/mod_perl/signed.cgi?IAXPhone Sorry for the 170 or so who have already signed. This list supposedly has 10,000 or more subscribers. 170 isn't very impressive. Please sign! ___ 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] (no subject)
Funny, they sell these old cards.. it seems like they are selling refurbs as new.. ... anyways RMA is on its way, would be nice if they would send one as a replacement first, so that we could continue our work and don't have to delay it. On Tue, 12 Apr 2005, Rich Adamson wrote: The only firmware upgrade procedure is for you to call digium support. Hi, I have just bought another TDM400P card from Digium directly, purchased last Thursday, received it today: Span 1: WCTDM/0 Wildcard TDM400P REV E/F Board 1 1 WCTDM/0/0 FXOKS (In use) 2 WCTDM/0/1 FXOKS (In use) 3 WCTDM/0/2 FXSKS (In use) 4 WCTDM/0/3 FXSKS (In use) So it seems that this is the same problem. I also find it strange that this card also has some intermittend issues with audio quality. We here are currently testing multiple different PBX solutions from Nortel BCM through Avaya and cisco to asterisk. Asterisk deffinitly has potential, though when cards have sound issues, that aint great. I hope Digium will send some sort of firmware upgrade procedure, if even possible. S. On Mon, 11 Apr 2005, Robert Webb wrote: Good morning all.. I was following a discussion on this list about the TDM400P revisions. It is my understanding that the current revision that one should have is the Rev. H and not the E/F. I have not yet been able to verify the rev stamped on the board, but zaptel is reporting that I have the Rev. E/F. I just bought this card in January direct from Digium and was wondering if I got the wrong Rev. somehow?? I have been having some intermittent problems but only thought it was my setup. ___ 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 ---End of Original Message- ___ 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] Getting CVS HEAD
Rich Adamson wrote: Hi, I want to download the CVS HEAD version. Any one can show how to get this version ? Is the version from: http://www.asterisk.org/index.php?menu=download the CVS Head version? How I can check if my version is CVS HEAD or not? phoenix*CLI show version Asterisk CVS-HEAD-04/07/05-11:36:47 I´ve downloaded it . Thanks all of you for the ideas and suggestions. The url you've shown above is correct. Just read the page. The # cvs checkout zaptel libpri asterisk is for cvs-head. The # cvs checkout -r v1-0 zaptel libpri asterisk is for v1 Stable. -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ 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] TE410P and X101P problem
Hi all Inewly added a X101P into my asterisk that already have a TE410P running 2 E1s namely span1 and span2 I am unable to get * to recognized the new X101P after i did modprbe wct4xxp and then modprobe wcfxo. ztcfg -vv reported all 63 channels are configured but zttool tells me that span 1,2,3 are OK and X101P UNCONFIGURED. I do not have anything plug into span 3 below are what i have zapata.conf [channels]context=defaultoverlapdial=nosignalling=pri_cpeswitchtype=euroisdnpridialplan=unknownrxwink=125echocancel=noechocancelwhenbridged=yesrxgain=0.0txgain=0.9immediate=yesmusiconhold=defaultgroup=1channel = 1-15,17-31busydetect=nogroup=2channel = 32-46,48-62busydetect=no group=5signalling=fxs_kschannel=63context=default zaptel.conf span=1,1,0,ccs,hdb3bchan=1-15,17-31dchan=16alaw=1-31 span=2,1,0,ccs,hdb3,crc4bchan=32-46,48-62dchan=47alaw=32-62 fxsks=63loadzone=us ztcfg -vv Channel 62: Individual Clear channel (A-law) (Slaves: 62)Channel 63: FXS Kewlstart (Default) (Slaves: 63) 63 channels configured. Do you Yahoo!? Make Yahoo! your home page ___ 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] From OH323 to SIP or OH323 without gatekeeper
Bruno Hertz wrote: Joe S [EMAIL PROTECTED] writes: Hi, I am new with asterisk. I was wondering if there is a way to call a OH323 user or SIP user using Netmeeting/SJPhone with H323 as the default protocol without having a gatekeeper. I can make a call from SIP to OH323 by specifying it in the extensions.conf file, like: exten=1001, 1, Dial(OH323/10.10.10.1) so I was wondering if there was a way to call from OH323 to SIP or OH323. Sure. Just specify in oh323.conf the context where incoming calls should go. That context then can include dial statements for any protocol, SIP, H323, IAX, whatever. See the Wiki for details on how to setup dial plans. Finally, instruct your H323 phone to use asterisk as a gateway resp. proxy, not a gatekeeper. Any calls will then go through asterisk, and to the context you specified. I'm doing that with Gnomemeeting all the time, and it works without problems. Mayabe can you show us a little sample? I can call from Gnomemeeting to Xlite, but no from xlite to gnomemeeting. Best regards, 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 -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ 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] Petition for IAX firmware
Wilson Pickett wrote: http://www.petitiononline.com/mod_perl/signed.cgi?IAXPhone I´ve signed before (in 90th posicion). Sorry for the 170 or so who have already signed. This list supposedly has 10,000 or more subscribers. 170 isn't very impressive. Please sign! ___ 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 -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ 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: Monitor with Asterisk@Home
Thank you for the reply.
exten = 1,1,SetVar(CALLFILENAME=${CALLERIDNUM})
exten = 1,2,SetVar(CALLTIME=${DATETIME})
exten = 1,3,SetVar(CALLPATH=/var/calls)
exten = 1,4,Monitor(wav,${CALLPATH}/${CALLTIME}-${CALLFILENAME},m)
? exten = 1,5,DIAL(SIP/something,15,t) - do I need to change SIP/something
exten = 1,6,StopMonitor
? exten = 1,7,Voicemail(u804) - what does u804 stand for or do
exten = 1,8,Hangup
exten = 1,102,StopMonitor
? exten = 1,103,VoiceMail(b804) -
exten = 1,104,Hangup
Would I also change
exten = 1,... to reflect the extention # if I am not using 1 as an
extension ie. Exten= 7726259 or
Do I put the above in the [ext-local] after each exten or does it get placed
in the
[ext-local]
include = ext-local-custom
exten = 7726257,1,Macro(exten-vm,[EMAIL PROTECTED],7726257)
exten = 7726258,1,Macro(exten-vm,[EMAIL PROTECTED],7726258)
exten = 7726259,1,Macro(exten-vm,[EMAIL PROTECTED],7726259)
exten = 9873022,1,Macro(exten-vm,[EMAIL PROTECTED],9873022)
exten = 9873023,1,Macro(exten-vm,novm,9873023)
or in the
[aa_1]
include = aa_1-custom
exten = 1,1,Goto(ext-local,7726258,1) ;
exten = 2,1,Goto(ext-local,7726259,1) ;
exten = 3,1,Goto(ext-local,7726257,1) ;
exten = fax,1,Goto(ext-fax,in_fax,1) ;
- snip -
Thankyou in return.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan
Sent: Monday, April 11, 2005 8:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Monitor with [EMAIL PROTECTED]
You have to put the monitor after the person presses their selection.
This is how ours is:
exten = s,1,answer
exten = s,2,SetCIDName('PMG')
exten = s,3,SetVar(company=PMG)
exten = s,4,Wait(1)
exten = s,5,DigitTimeout,5
exten = s,6,ResponseTimeout,40
exten = s,7,Background(/var/lib/asterisk/sounds/greetings/pmg)
exten = s,8,Background(greetings/dial)
exten = 1,1,SetVar(CALLFILENAME=${CALLERIDNUM})
exten = 1,2,SetVar(CALLTIME=${DATETIME})
exten = 1,3,SetVar(CALLPATH=/var/calls)
exten = 1,4,Monitor(wav,${CALLPATH}/${CALLTIME}-${CALLFILENAME},m)
exten = 1,5,DIAL(SIP/something,15,t)
exten = 1,6,StopMonitor
exten = 1,7,Voicemail(u804)
exten = 1,8,Hangup
exten = 1,102,StopMonitor
exten = 1,103,VoiceMail(b804)
exten = 1,104,Hangup
Kyle
mr. barker wrote:
I am sure that this was answered somewhere but my lack of being able
to find an answer using google I turn to the pros.
What would be the easist way to record all conversations using Monitor
command with the latest [EMAIL PROTECTED] ?
Using a FXO card with SIP extensions
I have tried adding the following in the extensions_additional.conf
but I am not getting a file generated in the
/var/spool/asterisk/monitor directory or anywhere else.
Help would be muchly appreciated.
Thanks for helping the newbiein return.
exten = s,7,Monitor(wav,${TIMESTAMP}-${CALLERIDNUM}-${MACRO_EXTEN})
[aa_1]
include = aa_1-custom
exten = 1,1,Goto(ext-local,7726258,1) ;
exten = 2,1,Goto(ext-local,7726259,1) ;
exten = 3,1,Goto(ext-local,7726257,1) ;
exten = fax,1,Goto(ext-fax,in_fax,1) ;
exten = h,1,Hangup();
exten = i,1,Playback(invalid) ;
exten = i,2,Goto(s,7);
include = ext-local
include = app-messagecenter
include = app-directory
exten = s,1,GotoIf($[${DIALSTATUS} = ANSWER]?4) ;
exten = s,2,Answer();
exten = s,3,Wait(1) ;
exten = s,4,SetVar(DIR-CONTEXT=default);
exten = s,5,DigitTimeout(3) ; Select
exten = s,6,ResponseTimeout(7) ;
exten = s,7,Monitor(wav,${TIMESTAMP}-${CALLERIDNUM}-${MACRO_EXTEN})
exten = s,8,Background(custom/aa_1) ; Press 1 for Peter Press 2 for
Paula Press 3 for the Kids
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[Asterisk-Users] Re: TE110P - NT-Mode ?
In article [EMAIL PROTECTED], Henry Jensen [EMAIL PROTECTED] wrote: Is there any way I can switch the TE110P card to NT-Mode ? In /etc/asterisk/zapata.conf, change signalling=pri_cpe to signalling=pri_net 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] Re: TE110P - NT-Mode ?
On Tue, Apr 12, 2005 at 12:23:52PM +, Tony Mountifield wrote: In article [EMAIL PROTECTED], Henry Jensen [EMAIL PROTECTED] wrote: Is there any way I can switch the TE110P card to NT-Mode ? In /etc/asterisk/zapata.conf, change signalling=pri_cpe to signalling=pri_net Wait a minute, you are saying, that pri_net is setting the card to NT mode? AFAIK NT mode must be set somewhere in the hardware configuration (e. g. when loading the kernel module - see http://www.voip-info.org/wiki-Asterisk+zaphfc). Nevertheless, I tried this already, the HiPath still says that it receives no signal and gives me a yellow alarm in zttool. Regards, 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] multiple asterisk boxes with show channels
If there are multiple asterisk boxes in use is there a way to "link" them together so when the manager api command "show channels" is executed ALL boxes are reported? Certainly I can connect to each box and execute the command show channels but was just wondering if there was something already in asterisk that I have not found yet that accomplishes this. THanks, Jerry ___ 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: From OH323 to SIP or OH323 without gatekeeper
Guillermo Salas M. [EMAIL PROTECTED] writes: Bruno Hertz wrote: Joe S [EMAIL PROTECTED] writes: Hi, I am new with asterisk. I was wondering if there is a way to call a OH323 user or SIP user using Netmeeting/SJPhone with H323 as the default protocol without having a gatekeeper. I can make a call from SIP to OH323 by specifying it in the extensions.conf file, like: exten=1001, 1, Dial(OH323/10.10.10.1) so I was wondering if there was a way to call from OH323 to SIP or OH323. Sure. Just specify in oh323.conf the context where incoming calls should go. That context then can include dial statements for any protocol, SIP, H323, IAX, whatever. See the Wiki for details on how to setup dial plans. Finally, instruct your H323 phone to use asterisk as a gateway resp. proxy, not a gatekeeper. Any calls will then go through asterisk, and to the context you specified. I'm doing that with Gnomemeeting all the time, and it works without problems. Mayabe can you show us a little sample? I can call from Gnomemeeting to Xlite, but no from xlite to gnomemeeting. Well, the direction GM - XLite basically was what we were talking about. For the other direction, i.e. calling an H323 client without gatekeeper, you simply dial the IP address or domain of the client, like Dial(OH323/yourclient.yourdomain.com:1720) or Dial(OH323/192.168.0.123:1720) somewhere in your Dialplan. E.g. if you want to do XLite - GM, such a dial statement should be part of the context into which your incoming SIP calls are routed, as specified in sip.conf. Example: * sip.conf context=default * extensions.conf [default] exten = 123,1,Dial(OH323/192.168.0.123:1720) I.e. dialing '123' with XLite registered on your server would in this case result in calling a hopefully running H323 client on IP address 192.168.0.123. Of course, if your H323 clients use dialup connections, setting up a dial plan for them without using a gatekeeper may prove to be troublesome. 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
[Asterisk-Users] Re: TE110P - NT-Mode ?
In article [EMAIL PROTECTED], Henry Jensen [EMAIL PROTECTED] wrote: On Tue, Apr 12, 2005 at 12:23:52PM +, Tony Mountifield wrote: In article [EMAIL PROTECTED], Henry Jensen [EMAIL PROTECTED] wrote: Is there any way I can switch the TE110P card to NT-Mode ? In /etc/asterisk/zapata.conf, change signalling=pri_cpe to signalling=pri_net Wait a minute, you are saying, that pri_net is setting the card to NT mode? AFAIK NT mode must be set somewhere in the hardware configuration (e. g. when loading the kernel module - see http://www.voip-info.org/wiki-Asterisk+zaphfc). Nevertheless, I tried this already, the HiPath still says that it receives no signal and gives me a yellow alarm in zttool. Sorry, I must have misunderstood. I thought you meant you wanted the card to behave as a switch rather than as a CPE. Perhaps instead, or as well, you need to use a crossover cable. That's what I have to use in order to connect one card port to another for testing. See http://www.voip-info.org/wiki-crossover+T1+cable 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] Remote phone often appears to be disconnected
Just set qualify=yes in sip.conf On Apr 12, 2005 3:41 AM, Ronald Wiplinger [EMAIL PROTECTED] wrote: Is there a possible settings for a remote SIP phone, so that a router will not close the connection due to long time inactivity? ___ 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] Changing DTMF mode depending on codec chosen
Thanks for writing back to me. Yep, just like you, I too am looking for a
lower bandwidth codec for my outbound. And, yes, broadvoice only officially
supports G.711. That being said, is there even a way to do this scenario in
asterisk?
Yes, there are frequently multiple ways to do things in asterisk.
Here's one example that was working before I discontinued their service.
(Note: since discontinuing their service, I deleted all of my old
extensions.conf entries, so I'm not able to include those below. This
particular * system was on a registered IP address with no firewall
or nat box; if it were behind a nat box, then additional statements
would be required for that.)
; register=myphonenum:[EMAIL PROTECTED] ; form #1
; register=myphonenum:[EMAIL PROTECTED]/1234; form #2
The register statement is _only_ needed to tell BV how to contact your
system for incoming calls. Without it, you won't get any incoming calls.
Notes for the two forms shown above include:
Form #1: required an entry in /etc/hosts like:
147.135.8.128 sip.broadvoice.com
Specifically note there is no parameter behind sip.broadvoice.com,
so all incoming calls will match exten=s in extensions.conf.
Form #2: The same statement but note the /1234 at the end. This
form requires an exten=1234 in the extensions.conf in order
to complete calls.
In the sip.conf section noted below, the type=friend is used as this
section was referenced for incoming calls (from BV), and for outgoing
calls to BV. (One could separate this section into type=user for incoming
calls, and type=peer for outgoing calls, and then specify different
parameters for each. There's no reason to do that since BV supports
only a very specific set of parameters for both incoming and outgoing
calls.)
; [broadvoice] ; this is referenced for outgoing calls to Broadvoice.com
; type=friend
; username=myphonenum
; secret=mysecret
; host=sip.broadvoice.com
; insecure=very
; canreinvite=no
; dtmfmode=inband
; fromuser=myphonenum
; fromdomain=sip.broadvoice.com
; context=from-broadvoice
; disallow=all
; allow=ulaw
; deny=0.0.0.0/0.0.0.0
; permit=147.135.8.129/255.255.255.0
; permit=147.135.0.129/255.255.255.0
; permit=147.135.4.128/255.255.255.0
Note that incoming calls are sent to the [from-broadvoice] context in
extensions.conf, however the incoming call is already negotiated with
allow=ulaw only. The deny and permit statements are there because you
don't know which of the various BV systems will actually be completing
calls to your * system, so I included all of them (that I could find
a few months ago). They may have added others by now, don't know.
The deny and permit statements are really there for basic security
purposes.
For outgoing calls, your extensions.conf would have an entry like:
[broadvoice-out]
exten = _1NX,1,SetCallerID(myDIDnum|a)
exten = _1NX,2,SetCIDName(myCallerIDname|a)
exten = _1NX,3,Dial(SIP/myphonenum:[EMAIL PROTECTED]/${EXTEN})
exten = _1NX,4,Congestion
The broadvoice keyword in the above refers back to the [broadvoice]
context in sip.conf (shown above). So when you make a call via BV, the
parameters in that context are used (including allow=ulaw and
dtmf=inband).
Again, keep in mind that I have deleted my extensions.conf entries, so
the above statements may have syntax errors, etc. I simply typed the
above from memory. Don't be cutting/pasting it into your system
without understanding what you're doing.
Other things to keep in mind about BV:
1. BV does not use asterisk for their switch (or if they do, its highly
modified). What you've learned about asterisk does _not_ necessarily
apply to their soft switch. Their switch may negotiate things differently
then *, etc.
2. One of the BV employees implemented asterisk at his home, got it to
work, and published the parameters he used to make it work. His
implementation (and published parameters) is specific to his system,
which no one knows whether he's on a registered IP, behind a nat box,
etc. So, his published parameters are _only_ a starting point, not a
firm recommendation that anyone could cut/paste. That's one of the
reasons why so many people have setup problems with asterisk (combined
with a lack of knowledge/experience as to how to diagnose problems).
3. BV had a problem with the way asterisk systems would re-register
every minute or two, and Olle wrote a patch for asterisk that reduced
that re-register traffic. (BV was being pounded by all of the remote
asterisk systems beating on their systems with that re-register traffic,
and threatened to discontinue everyone's service that continued to do that.
That patch only applied to those systems that were located behind a nat
box, but the patch didn't damage anything if you had a registered IP
address. I believe the patch made it into both cvs-head and stable.)
4. One of the reasons why BV doesn't
Re: [Asterisk-Users] (no subject)
Funny, they sell these old cards.. it seems like they are selling refurbs as new.. ... anyways RMA is on its way, would be nice if they would send one as a replacement first, so that we could continue our work and don't have to delay it. They can, its called cross-shipment, but they need a credit card number to ensure they get your return shipment. You have to ask for it. That's the way I did it. Regarding the refurbs, if you or I were owners of digium, how would we handle a backstock of older (possibly refurb) cards when its somewhat known the old cards work fine in some systems? (And, we don't have a clue which systems/motherboards the cards worked fine in.) ___ 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] Interface bonding + asterisk
On Monday 11 April 2005 15:15, Jesus Mogollon wrote: Hi all I installed asterisk on a dual PIII 700 with two NICs. I then proceeded to configure both NICs with bonding enable (bonding miimon=100 mode=1). I know certain features (like load balancing) under a bonded configuration is not understood by some switches, so I configured it using mode=1 (Failover only). The problem I'm having is that, sometimes, calls start fine but then one of the parties loses audio (it could be the caller of the callee who loses audio, there is no pattern). I was wondering if someone has hit the same wall as me. There are people using this server right now, so I haven't tried the no-bonding option as it means downtime. Any help would be appreciated. I've never tried bonding, but if I was using multiple interfaces, I'd use a simple trick routers use and configure a loopback address, ensure the routing table propagates it and have * listen on only that i/f. Try, ifconfig lo:1 some IP address netmask 255.255.255.255 B ___ 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 get list of codecs
Hi Guys, Is it possible to get the UAC supported codec list when making a call. I want to assign to variable1 and variable2 the first 2 supported codecs using AGI script e.g. $variable1=g723 $variable2=g729 Somebody can help me ? Any help is appreciated. Thanks, Pavel Siderov ___ 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] Low cost box for hosting Asterisk and atleastone TDM400p - THIN CLIENT MAYBE?
Hi, Actually I guess what I am looking for is semi-sealed box that I can add 1 or 2 PCI cards too. A regular PC work work in most cases since I do not want a keyboard or mouse attached to it. I do not want users screwing with the system. If it is sealed with no monitor/keyboard/mouse then they can't screw it up very easily. I guess I am looking for something that is somewhere in between a PC and Linksys router box. One possibility might be a thin client box, but I haven't found any sources for an OEM box. I looked at the HP (http://h18004.www1.hp.com/products/thinclients/index_t5000.html) thin clients but I can get a Dell Box for the same price that does more. 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 Addons compile errors
Try re-downloading Asterisk-Addons. It sounds like you have the version that is meant for CVS HEAD and not the stable 1.0 series. -mishehu lie ka wrote: HI: I have compiled and installed Asterisk 1.0.7 without any problems.I have also installed mysql and DBD::mysql successfuly / When I tried to make asterisk-addons, it showed me the problem like these: [EMAIL PROTECTED] asterisk-addons]# make install ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` app_addon_sql_mysql.c:162:64: macro AST_LIST_REMOVE requires 4 arguments, but only 3 given make -C format_mp3 all make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o common.o common.c gcc -pipe -fPIC -Wall -WstricUntitled 1t-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o dct64_i386.o dct64_i386.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o decode_ntom.o decode_ntom.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o layer3.o layer3.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o tabinit.o tabinit.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o interface.o interface.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o format_mp3.o format_mp3.c format_mp3.c: In function `load_module': format_mp3.c:335: warning: passing arg 5 of `ast_format_register' from incompatible pointer type gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -shared -Xlinker -x -o format_mp3.so common.o dct64_i386.o decode_ntom.o layer3.o tabinit.o interface.o format_mp3.o make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3' cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_saycountpl.o app_saycountpl.c cc -shared -Xlinker -x -o app_saycountpl.so app_saycountpl.o cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c: In function `my_load_module': cdr_addon_mysql.c:269: warning: assignment makes pointer from integer without a cast cc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o -lmysqlclient -lz -L/usr/lib/mysql cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:162:64: macro AST_LIST_REMOVE requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:162: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:162: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:162: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 I am not a Linux Expert. What can I do for make addons? Thank to all, and sorry for my poor english. _ Do You Yahoo!? http://cn.rd.yahoo.com/mail_cn/tag/1g/*http://cn.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 !DSPAM:425b96c05442027214946! ___ 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] Multiple Servers and 1 Central Voicemail
But voicemailboxes have to exists on all asterisk servers right? Also, what happens if for example, the user is accessing his VMB on server 1 and changes his password, then travel to where server 2 is and tries to access his VMB? the config on server2 would still have the old one so you need to sync voicemail.conf on all servers too ... From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason BrownSent: Lunes, 11 de Abril de 2005 07:19 a.m.To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Multiple Servers and 1 Central Voicemail MWI works just fine. ___ 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] Version 0.80 of IPS released
Version 0.80 - 12. April 2005. You spit out the versions faster than I can reinstall them :) Did you by any chance had the time to take a look a transfer problem when there are two active calls on a monitored extension ? Ivan ___ 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] Low cost box for hosting Asterisk and atleastoneTDM400p - THIN CLIENT MAYBE?
Depending on how many users you want to support and price, there are lots of options. Smallest form factor will be SOC (System on Chip) These are little more costly and not going to carry a huge load. Next would be Mini-ITX A bit bigger and will carry more load. VIA is the king in this arena though you can find some amazing parts that are based upon the Pentium M now. Micro-ATX and up after that. Cheap and small are not really synonymous... There are several wall mount types of boxes if you want to really secure a PC in a no monitor/no mouse case. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn Sent: Tuesday, April 12, 2005 6:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Low cost box for hosting Asterisk and atleastoneTDM400p - THIN CLIENT MAYBE? Hi, Actually I guess what I am looking for is semi-sealed box that I can add 1 or 2 PCI cards too. A regular PC work work in most cases since I do not want a keyboard or mouse attached to it. I do not want users screwing with the system. If it is sealed with no monitor/keyboard/mouse then they can't screw it up very easily. I guess I am looking for something that is somewhere in between a PC and Linksys router box. One possibility might be a thin client box, but I haven't found any sources for an OEM box. I looked at the HP (http://h18004.www1.hp.com/products/thinclients/index_t5000.html) thin clients but I can get a Dell Box for the same price that does more. 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 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] NENA CAMA Trunks for 911 and *
Has anyone ever explored what would be required to enable * to produce NENA standard CAMA signaling for interconnection with conventional e911 services? ___ 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] Cannot open chan_zap:
On Monday 11 April 2005 22:36, Tim Connolly wrote: I'm assuming I would see an error if this was bad: ldd /usr/lib/asterisk/modules/chan_zap.so linux-gate.so.1 = (0xe000) libpri.so.1 = /usr/lib/libpri.so.1 (0xb7f89000) libtonezone.so.1.0 = /usr/lib/libtonezone.so.1.0 (0xb7f68000) libc.so.6 = /lib/tls/libc.so.6 (0xb7e3f000) libm.so.6 = /lib/tls/libm.so.6 (0xb7e1c000) /lib/ld-linux.so.2 (0x8000) Still, Unable to load module chan_zap Apr 11 16:37:04 WARNING[21531]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_zap: cannot open shared object file: No such file or directory Have * load chan_zap automatically then run strace -f -o asterisk.out asterisk -gvc Study the output file asterisk.out. B ___ 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] Meetme disconnecting clients that use VAD
Hi there I am using Meetme and am connecting with clients that use VAD. The clients have been built with RTC Client API. What Meetme seems to do is cut users off from the conference if it does not receive any audio packets from the user for 1 minute 45 seconds. The solution I have found to this problem is to disconnect and reconnect clients every minute or so, but this solution is not ideal. Is there any way to configure Meetme so that it does not disconnect clients in this way? Many thanks Steven ___ 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] Internet Conection Broken and asterisk can not route any calls
Hello all, Sometimes my ADSL internet conection, gets down and I cannot access to internet. When this happens, my asterisk gets crazy and it cannot route my calls. Actualy I have an scape secuence (111) followed with the PSTN number, and the call is routed trought my ISDN lines. When my ADSL gets down, I cannot make any calls. If I dial 0 followed the PSTN number I will use an VoIP provider trought Internet. I know that if there is no internet I can not make VoIP calls, but why I can not make ISDN either? Any clue will be welcomed. Thanks, Ismael. ___ 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] VAD/DTX implementation through zaptel cards
Eric Wieling wrote: [EMAIL PROTECTED] wrote: Hi, How can i implement VAD/DTX using zaptel with asterisk towards PSTN. TDM (PSTN/telcos) do not support VAD. The entire idea of VAD is not even a valid idea. Doing VAD on audio coming _from_ the TDM world certainly is something you might want to do, to dramatically reduce the bandwidth you consume when sending the audio via VoIP channels. This kind of thing is not presently implemented in *, though, but it could be. (note: doing it well will require a bunch of CPU, though. I wonder if it could be done in the same DSP that is doing echo-cancellation on the new TE4xxP boards? -SteveK ___ 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] Getting CVS HEAD
Have you actually tried that cvs-up script? Not knowing how to check my version and using that cvs-up thing caused me a lot of grief. I thought I was on CVS-HEAD when I was on um.. something else that didn't really have a version in show version. Andy Hamilton wrote: The fastest way to obtain Asterisk is to use CVSup. To check out Asterisk using CVSup, create a sup file as follows: *default host=cvs.digium.com *default base=/usr/src *default release=cvs tag=. *default delete use-rel-suffix asterisk libpri zaptel Perhaps call it asterisk-sup and put it in /usr/src Then simply: # cd /usr/src # cvsup asterisk-su[ ___ 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] multiple line usage on Polycom IP300
can anyone help ?? trying to get Polycom IP300 to utilize both lines, would like calls to rollto open linewhen incoming call arrives while user is on line 1. Looked everywhere and tried many things with no luck. ___ 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] TE410P and X101P problem
LeeLee- Try configuring all 4 spans first and then the single channel (125) above that - works for me. Modprobe in the same order, then ztcfg. Regards Scott Stingel www.evtmedia.com Lee Lee wrote: Hi all I newly added a X101P into my asterisk that already have a TE410P running 2 E1s namely span1 and span2 I am unable to get * to recognized the new X101P after i did modprbe wct4xxp and then modprobe wcfxo. ztcfg -vv reported all 63 channels are configured but zttool tells me that span 1,2,3 are OK and X101P UNCONFIGURED. I do not have anything plug into span 3 below are what i have _zapata.conf_ [channels] context=default overlapdial=no signalling=pri_cpe switchtype=euroisdn pridialplan=unknown rxwink=125 echocancel=no echocancelwhenbridged=yes rxgain=0.0 txgain=0.9 immediate=yes musiconhold=default group=1 channel = 1-15,17-31 busydetect=no group=2 channel = 32-46,48-62 busydetect=no group=5 signalling=fxs_ks channel=63 context=default _zaptel.conf_ span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 alaw=1-31 span=2,1,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 alaw=32-62 fxsks=63 loadzone=us _ztcfg -vv_ Channel 62: Individual Clear channel (A-law) (Slaves: 62) Channel 63: FXS Kewlstart (Default) (Slaves: 63) 63 channels configured. Do you Yahoo!? Make Yahoo! your home page http://us.rd.yahoo.com/my/navbar/sethp/*http://www.yahoo.com/r/hs ___ 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] Trunk Seize - Line 1 - CO1: Does it exist in an Asterisk environment?
Check out call parking. It's basically the same thing. -mishehu Ben Ryan wrote: I have a question probably for those in the know in business Asterisk solutions. I have searched high and low but have not been able to get any answers. I hope there is someone on the list that can answer my question. How do you implement trunk seize? This is a feature that is almost universal in the conventional PBX world. Say a user, Jane takes a call. The call is for someone else - Fred. Jane knows Fred is often not in his office, therefore can't do an extension transfer. Jane hits HOLD, puts a page out for the recipient Fred, telephone call on Line 1, Fred is out in another office and takes the call by picking up the handset and hitting CO1/Line1. How does this happen in a VoIP environment - in Asterisk? What about other IP telephony environments? And crucially, with multiple inbound lines, how do you determine a specific line to grab? Thanks for any illumination you can provide. ___ 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 !DSPAM:425b4032302625654620869! ___ 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] Problem with fxo
I have no Idea of the strange errors, but as far as i know, the proper
way of calling is:
Zap/g${group}/${phone_number}
where ${group} is a valid group inside zapata.conf, and
${phone_number} is the desired PSTN phone to call. In you email you
wrote the messages and i can see that you missed the letter 'g'
before the group and the last '/' slash. Give that a try, may be will
work.
Best Regards
- Moy
On Apr 12, 2005 11:23 AM, Julio Saura [EMAIL PROTECTED] wrote:
Hi,
i am trying to use my fxo card for analog calls ..
fxo card seems to be ok, working properly but when trying to call
outside ( from a sip phone ot pstn ) i get the following error on
asterisk .
Apr 12 11:59:24 DEBUG[4231]: chan_sip.c:4633 build_route: build_route:
Contact hop: Drugo sip:[EMAIL PROTECTED]:5060
-- Executing Dial(SIP/69-562c, Zap/1/651559526|5) in new stack
Apr 12 11:59:24 DEBUG[4231]: chan_zap.c:1645 zt_call: Dialing
'651559526'
Apr 12 11:59:24 DEBUG[4231]: chan_zap.c:1706 zt_call: Deferring
dialing...
-- Called 1/651559526
Apr 12 11:59:25 DEBUG[4231]: chan_zap.c:3770 __zt_exception: Exception
on 15, channel 1
Apr 12 11:59:25 DEBUG[4231]: chan_zap.c:3082 zt_handle_event: Got event
Hook Transition Complete(12) on channel 1 (index 0)
Apr 12 11:59:27 DEBUG[4231]: chan_zap.c:3770 __zt_exception: Exception
on 15, channel 1
Apr 12 11:59:27 DEBUG[4231]: chan_zap.c:3082 zt_handle_event: Got event
Dial Complete(9) on channel 1 (index 0)
Apr 12 11:59:27 DEBUG[4231]: chan_zap.c:1224 zt_enable_ec: Enabled echo
cancellation on channel 1
Apr 12 11:59:27 DEBUG[4231]: channel.c:1363 ast_read: Dropping duplicate
answer!
any clue?
got no info about exception 15 :/
Thanks in advance
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Re: [Asterisk-Users] TE410P and X101P problem
On Tue, 2005-04-12 at 05:14 -0700, Lee Lee wrote: Hi all I newly added a X101P into my asterisk that already have a TE410P running 2 E1s namely span1 and span2 I am unable to get * to recognized the new X101P after i did modprbe wct4xxp and then modprobe wcfxo. ztcfg -vv reported all 63 channels are configured but zttool tells me that span 1,2,3 are OK and X101P UNCONFIGURED. I do not have anything plug into span 3 Even if you don't config spans 3/4 you still need to leave space for them. So, if you are using E1 (sounds like it) the x101p should be 125 not 63. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] Version 0.80 of IPS released
On Tue, 2005-04-12 at 13:40 +0200, Thorben Jensen wrote: Version 0.80 - 12. April 2005. * Swedish language added - thanks Daniel Nylander * Bug fixes Any chance of integrating some sort of input text box, where you can just type in the extension number and hit enter to transfer a call? Maybe with some option for attended transfer as well even? Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] multiple line usage on Polycom IP300
On Tuesday 12 April 2005 10:18 am, MobilPete wrote: can anyone help ?? trying to get Polycom IP300 to utilize both lines, would like calls to roll to open line when incoming call arrives while user is on line 1. Looked everywhere and tried many things with no luck. Do you have your lines register sepratly? E.g. is there a seperate entry in sip.conf for each line or do they both register as the same sip device? -- 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] Power Consumption of a Digium Wildcard TE410P
Hi *, Does anyody know, what power consumption this card have? The technical descripten is really quiet at this point .. Tfh, Oliver ___ 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] VAD/DTX implementation through zaptel cards
Steve Kann wrote: Eric Wieling wrote: [EMAIL PROTECTED] wrote: Hi, How can i implement VAD/DTX using zaptel with asterisk towards PSTN. TDM (PSTN/telcos) do not support VAD. The entire idea of VAD is not even a valid idea. Doing VAD on audio coming _from_ the TDM world certainly is something you might want to do, to dramatically reduce the bandwidth you consume when sending the audio via VoIP channels. This kind of thing is not presently implemented in *, though, but it could be. (note: doing it well will require a bunch of CPU, though. I wonder if it could be done in the same DSP that is doing echo-cancellation on the new TE4xxP boards? Unless Digium's plans changed since the last time I spoke to Mark, the answer would be no. I believe they are using a dedicated function echo canceller device. Regards, Steve ___ 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: Ebay listing selling Asterisk @ Home andAMPfor over 1000 dollars
It is alright to sell hardware, and it is alright to sell labor when dealing with open source software. But selling licensing on something that does not exist (extension licensing???) is wrong. What if somebody started charging extra licensing to use the include music tracks for MOH? Also, I do believe that if Digium and the other software copyright holders are concerned about not receiving the due credit for any purported use of their software in this ebay listing, I do believe there are legal remedies available. -mishehu SNIP Email Trail history Simple questionso what? It may be easier to setup to you, but it is mind boggingly difficult for others. Extensions, Ring Groups, Trunks, Dial Plans, PSTN, POTS, PRI, etc can start to make many people's eyes gloss over. That is the beauty of this business. I am in independent consultant, do you feel it is wrog for me to sell an Asterisk system to a client to solve their business needs? Who knows, I might have started selling pre-configured boxes on eBay myself except looking through completed items shows that while he has been selling that box for sometime, he has never had a single bid on one. And its NOT because its based on open source, PBX's in general do not sell very well if at all on eBay as it is a function of the phone guy to recommend a system to his clients. With Asterisk, the phone guy can now be the computer guy and can handle both systems. But anyone that thinks it is immoral in some way for me to make a living off if it is crazy. How many Linux admins are employed out there? Shouldn't they give their time away because Linux is open source? What about Dell selling servers with Linux installed? Should they give away the servers? So why should a phone system be any different? Kerry ___ 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] Dialing Out
On Apr 11, 2005 8:11 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: nat=no disallow=all allow=g729 allow=g726 auth=plain context=default canreinvite=yes username=USERNAME secret=PASSWORD dtmfmode=info fromdomain=REALM fromuser=USERNAME qualify=1000 insecure=very I am using Asterisk 1.0.7 compiled into RPMs from the tarball running on CentOS Enterprise 4... Can someone point me in the right direction... Doug You haven't stated what your problem is. ___ 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] DS3000P - 20 E1 capacity on single card
Andres wrote: Can you confirm if there will be some sort of DSP daughther card add on of some sort for the DS3000 so that we can run G729 transcoding? I don't see how the DS3 interface would be usefull unless we could offload transcoding stuff to onboard DSPs. Or is Digium only going to recommend this card for G711 only uses? No, I cannot comment on that. It is safe to say that for non-transcoding applications, any reasonable 64-bit CPU should be able to handle the full traffic load of a DS-3. A 32-bit CPU will run into problems supporting an adequate number of threads. ___ 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] DS3000P - 20 E1 capacity on single card
Andrew Kohlsmith wrote: secondary card for DSP functions is very inefficient of the PCI bus. I'd be curious to know if the Digium cards can even do PCI-PCI DMA. The Digium TDM cards can DMA into any RAM accessible over the PCI bus, regardless of whether it is located on the motherboard or on a PCI card. ___ 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] Acceptable voice time delay
What is considered an acceptable time delay between two servers for a fair (not neccessarily great) voice quality. I use voipjet to connect my calls from iax2 to the pstn. Although the sound quality is good, there is considerable time delay, I wait seconds before the other party hear what I say. It becomes more of a walkie talk. When I ping voipjet, it takes about 600ms. would i take it that the extra dely is coming from the voipjet server to the pstn and i should try other servers or is there some other issues. I am behind a nat, would that make any difference? Can you provide me with Europe based iax termiantion other than voiptalk. I appreciate some of you guys telling me the ms ping to your providers so i can have an idea. Thanks a bunch. __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ 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] Has anyone got Asterisk working behind a NAT connection to users within a NAT
On 11:27, Tue 12 Apr 05, Wilson Pickett wrote: However, and I know this is a running issues, I cannot get external sip users behind a NAT to be able to successfully connect to asterisk when it's behind a NAT as well. I have done port forwarding at both ends dealing with the usual ports of 5060, 4569 and 5036 as well as opening up the rtp ports for the voice traffic on 10,000 to 20,000. Is there a way without asterisk being on an external ip? Are you using nat=yes in sip.conf entries and giving the externip and localnet parameters? Also set the canreinvite=no for the external phones. that way the audio stream is always managed thru * -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ 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] multiple line usage on Polycom IP300
If you have two lines registered to one phone then you need to do the following... This assumes extensions 1001 and 1002 are your line appearances... exten = 1001,1,Dial(1001,20,trf) ;we are dialing line 1 -- After 20 seconds it will timeout and go to the next line exten = 1001,2,Dial(1002,20,trf) ;just told it to dial line 1002 exten = 1001,3,Do your voice Mail Here exten = 1001,4,Hangup You could alternately just use a GoTo after the 1st dial attempt times out and send the call to 1002 If you are talking about getting a second call while on line 1, then you just need to enable call waiting on the Asterisk box. The phone should automatically show a second incoming call and allow you to place call 1 on hold. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josiah Bryan Sent: Tuesday, April 12, 2005 7:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] multiple line usage on Polycom IP300 On Tuesday 12 April 2005 10:18 am, MobilPete wrote: can anyone help ?? trying to get Polycom IP300 to utilize both lines, would like calls to roll to open line when incoming call arrives while user is on line 1. Looked everywhere and tried many things with no luck. Do you have your lines register sepratly? E.g. is there a seperate entry in sip.conf for each line or do they both register as the same sip device? -- 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 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] Acceptable voice time delay
chawki hammoud wrote: What is considered an acceptable time delay between two servers for a fair (not neccessarily great) voice quality. I can't really deal with anything over 150ms, although regular users will tolerate ~200ms. I use voipjet to connect my calls from iax2 to the pstn. Although the sound quality is good, there is considerable time delay, I wait seconds before the other party hear what I say. It becomes more of a walkie talk. When I ping voipjet, it takes about 600ms. There's your problem. 600ms stinks. Thanks a bunch. No problem. I don't know if voicepulse can do Europe iax term, but it's worth looking into. I've had pretty good experiences with them so far ( excepting the price hike...but what can you do? ). Sean ___ 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: Ebay listing selling Asterisk @ Home (Blah Blah)
Message: 14 Date: Mon, 11 Apr 2005 17:35:05 -0400 From: dean collins [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Lol, just posted a question to the list that should keep away any bidders. [Christopher Jacob] Why? Is there some reason this person shouldn't make a living selling Asterisk / AMP / FOP etc??? In fact, he is at least fessing up to that fact that it is Asterisk AND open source. While he of course has to include all source (or provide access to it) he doesn't have to advertise the fact that it is Asterisk. Thankfully, people all over the world are selling Asterisk and Asterisk related services. It's what gives the product a foot hold. It's what finances digium. Do you think that the guy that developed AMP did it without intending to make some cash off of it? The released it as OSS, (which is awesome) but of course they are going to continue to sell it. Freak. ~c ___ 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] Line Noise HELP!
Thanks for that Rich. Etheral trace is going to be almost impossible for various reasons, but will try the other two options. Can't find much online re. debugging - any chance of killing the box by turning this on? SIP show channels and the various CAPI show commands do not show anything out of the ordinary when the problem occurs. In order for anyone to help identify the noise problem, you really are going to have to find a way to capture some data, otherwise we're all spinning our wheels and guessing. To implement debugging, look at /etc/asterisk/logger.conf and add the keyword 'debug' like: messages = notice,warning,error,debug Adding that keyword requires that * be stopped and restarted to take effect. That tells asterisk to log all debug statements (that are embedded in asterisk source code) to write to /var/log/asterisk/debug file. That debug file will grow to a very large size rather quickly, so you need to pay attention to available disk space, etc. When the noise problem occurs, note the specific system time, and take a look at /var/log/asterisk/debug to see what was happening around that time. Once you've captured at least some data, you may want to remove the debug statement. If you haven't tried some of the other cli debug tools, you might want to do help sip debug, help rtp debug, etc. If you can't run ethereal on the system with the problem, there are other tools like tcpdump, etc, that can be used to capture packets. ___ 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] Petition for IAX firmware
Sign it: http://www.petitiononline.com/mod_perl/signed.cgi?IAXPhone Now it would be even more interesting to see if Cisco or maybe Siemens/Polycom would bring out a firmware for IAX, now that would be a revolution.. :) Cisco et al won't exactly be blown away by the not even 200 sigs :) I wonder how Star Trek got back on the air with fan petitions? ___ 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] multiple line usage on Polycom IP300
we tried both, setting it as same and also seperate. but niether worked. - Original Message - From: Josiah Bryan [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 12, 2005 9:41 AM Subject: Re: [Asterisk-Users] multiple line usage on Polycom IP300 On Tuesday 12 April 2005 10:18 am, MobilPete wrote: can anyone help ?? trying to get Polycom IP300 to utilize both lines, would like calls to roll to open line when incoming call arrives while user is on line 1. Looked everywhere and tried many things with no luck. Do you have your lines register sepratly? E.g. is there a seperate entry in sip.conf for each line or do they both register as the same sip device? -- 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 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] Acceptable voice time delay
Around 250ms max. Over that and you will have the walkie-talkie effect you are experiencing. So with you 600ms delay you are way over the top. There is also the delay on the call on the PSTN side you have to take into account. For example, I am in Europe and making a call to the UK via Voipjet is usually OK. But making a call to Romania is a lottery. Sometimes it is great, sometimes the delay is huge. And that has nothing to do with the internet delay which is more or less constant at around 180ms for me. But with 600ms, which is over half a second, you have problems. ___ 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: multiple line usage on Polycom IP300
On Tuesday 12 April 2005 10:18 am, MobilPete wrote: can anyone help ?? trying to get Polycom IP300 to utilize both lines, would like calls to roll to open line when incoming call arrives while user is on line 1. Looked everywhere and tried many things with no luck. Do you have your lines register sepratly? E.g. is there a seperate entry in sip.conf for each line or do they both register as the same sip device? Yes, a good way to do it is to register each line separately, like this: sip.conf [100] type=friend username=100 secret=100 callerid=100 host=dynamic dtmfmode=rfc2833 context=extensions_context [EMAIL PROTECTED] disallow=all allow=ulaw [100b] type=friend username=100b secret=100b callerid=100 host=dynamic dtmfmode=rfc2833 context=extensions_context disallow=all allow=ulaw Then, you can use SetGroup and CheckGroup like this in your dialplan to bypass the annoying call waiting feature: extensions.conf exten = 100,1,SetGroup(100) exten = 100,2,CheckGroup(1) exten = 100,103,Goto(100b,1) exten = 100,3,Dial(SIP/100,20) exten = 100,4,Voicemail(su100) exten = 100,5,Hangup exten = 100b,1,Dial(SIP/100b,20) exten = 100b,2,Voicemail(sb100) exten = 100b,3,Hangup exten = 100b,102,Voicemail(sb100) exten = 100b,103,Hangup - Noah ___ 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] Version 0.80 of IPS released
Ronald Wiplinger wrote: Adam Goryachev wrote: On Tue, 2005-04-12 at 13:40 +0200, Thorben Jensen wrote: Version 0.80 - 12. April 2005. * Swedish language added - thanks Daniel Nylander * Bug fixes Any chance of integrating some sort of input text box, where you can just type in the extension number and hit enter to transfer a call? Maybe with some option for attended transfer as well even? Regards, Adam Some others, would it be possible to pass a message to a phone (onto the display) ??? On the Calls tab the Caller ID Name is for incoming calls for me correct, since caller-id on my Zap device does not work with unknown The list of outgoing calls shows an empty space, I believe it should show the same caller-id, as it does on the Panel tab. 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] Remote phone often appears to be disconnected
Julian J. M. wrote: Just set qualify=yes in sip.conf This I have already, but does not help. I believe it is the ADSL router at the remote end, which may disconnect due to inactivity. I think I can change the ttl parameter on the phone, but than I have to go there. I was looking for something that I can do that from the server end. bye Ronald On Apr 12, 2005 3:41 AM, Ronald Wiplinger [EMAIL PROTECTED] wrote: Is there a possible settings for a remote SIP phone, so that a router will not close the connection due to long time inactivity? ___ 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 -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ 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: Asterisk-Users Digest, Vol 9, Issue 104
[EMAIL PROTECTED] is believed to have said: http://www.petitiononline.com/mod_perl/signed.cgi?IAXPhone Sorry for the 170 or so who have already signed. This list supposedly has 10,000 or more subscribers. 170 isn't very impressive. Please sign! Just signed; more hardware side support to the IAX protocol can only be a good thing. I hope more signatures will arrive! Rgds Aldo ___ 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 get list of codecs
mmm i think Agi by itself does not provide a way to do so. And the codecs are negotiated depending upon the codec that both call sides support. So, i belive that the only way is making your own implementation of AGI in res_agi.c :) Hopefully someone will come up with a better idea :-) best regards On Apr 12, 2005 1:34 PM, Pavel Siderov - Hostmates [EMAIL PROTECTED] wrote: Hi Guys, Is it possible to get the UAC supported codec list when making a call. I want to assign to variable1 and variable2 the first 2 supported codecs using AGI script e.g. $variable1=g723 $variable2=g729 Somebody can help me ? Any help is appreciated. Thanks, Pavel Siderov ___ 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] DS3000P - 20 E1 capacity on single card
[EMAIL PROTECTED] wrote: Andres wrote: Can you confirm if there will be some sort of DSP daughther card add on of some sort for the DS3000 so that we can run G729 transcoding? I don't see how the DS3 interface would be usefull unless we could offload transcoding stuff to onboard DSPs. Or is Digium only going to recommend this card for G711 only uses? No, I cannot comment on that. Kevin, What is target release date for DS3000P? ___ 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] multiple line usage on Polycom IP300
Polycom enables call waiting on each line button. If you wish the second call to go directly to the second button you need o keep track of this with group in * and control with your dial plan. On Apr 12, 2005, at 9:41 AM, Josiah Bryan wrote: On Tuesday 12 April 2005 10:18 am, MobilPete wrote: can anyone help ?? trying to get Polycom IP300 to utilize both lines, would like calls to roll to open line when incoming call arrives while user is on line 1. Looked everywhere and tried many things with no luck. Do you have your lines register sepratly? E.g. is there a seperate entry in sip.conf for each line or do they both register as the same sip device? -- 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 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: Ebay listing selling Asterisk @Home andAMPfor over 1000 dollars
This guy is not selling extension licensing, he is selling a pre-configured system and charges extra to configure more extensions and says other people charge extra licenses for extensions of which I can only find the big PBX manufacturers that do that. Regardless, you sure can charge extension licensing if you choose to and it is a common practice to do that as a way of creating a service contract for your client. A typical extension license provides the programming, support, and maintenance of that extension. There is absolutely nothing wrong with that. The music on hold music is from FreePlayMusic whose licensing agreement on their site would make the use of that music within Asterisk a violation of their licensing agreement. I haven't looked in Asterisk licensing to find where the use of the tracks from FreePlayMusic are mentioned, and without that, they have every right to charge for them. The only difference between deploying an Asterisk server and a legacy style PBX is the cost of the underlieing technology. The rest of the business model still applies. Finally, isn't this a whole lot of nothing anyway since he has never had even 1 bid on any of his previous auctions for the same thing? -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of I put the Who? in Mishehu Sent: Tuesday, April 12, 2005 7:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RE: Ebay listing selling Asterisk @Home andAMPfor over 1000 dollars It is alright to sell hardware, and it is alright to sell labor when dealing with open source software. But selling licensing on something that does not exist (extension licensing???) is wrong. What if somebody started charging extra licensing to use the include music tracks for MOH? Also, I do believe that if Digium and the other software copyright holders are concerned about not receiving the due credit for any purported use of their software in this ebay listing, I do believe there are legal remedies available. -mishehu SNIP Email Trail history Simple questionso what? It may be easier to setup to you, but it is mind boggingly difficult for others. Extensions, Ring Groups, Trunks, Dial Plans, PSTN, POTS, PRI, etc can start to make many people's eyes gloss over. That is the beauty of this business. I am in independent consultant, do you feel it is wrog for me to sell an Asterisk system to a client to solve their business needs? Who knows, I might have started selling pre-configured boxes on eBay myself except looking through completed items shows that while he has been selling that box for sometime, he has never had a single bid on one. And its NOT because its based on open source, PBX's in general do not sell very well if at all on eBay as it is a function of the phone guy to recommend a system to his clients. With Asterisk, the phone guy can now be the computer guy and can handle both systems. But anyone that thinks it is immoral in some way for me to make a living off if it is crazy. How many Linux admins are employed out there? Shouldn't they give their time away because Linux is open source? What about Dell selling servers with Linux installed? Should they give away the servers? So why should a phone system be any different? Kerry ___ 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] Dialing Out (My mistake, here is the entire message)
Sorry about before, I sent the message from the wrong address and didn't repaste the entire message when I sent it from the right address... I am having a problem using my backup dialout termination from asterisk. The server I am registered to for back up is running SER 0.90. If I dial NUMBER1, which is locally registered number on the SER server, it goes through. However, if I dial NUMBER2 which SER should forward to it's termination provider, I get the following error from Asterisk... I am registered with the SER server, so I know my password is right... Apr 11 16:11:23 WARNING[10647]: chan_sip.c:6829 handle_response: Forbidden - wrong password on authentication for INVITE to 'ATAADAPTER sip:[EMAIL PROTECTED];tag=as4a0d4cf3' Here are my sip.conf sections for the SER server... [SerServer] type=friend host=IP GOES HERE nat=no disallow=all allow=g729 allow=g726 auth=plain context=default canreinvite=yes username=USERNAME secret=PASSWORD dtmfmode=info fromdomain=REALM fromuser=USERNAME qualify=1000 insecure=very I am using Asterisk 1.0.7 compiled into RPMs from the tarball running on CentOS Enterprise 4... Can someone point me in the right direction... Doug ___ 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] Low cost box for hosting Asterisk and atleastone TDM400p - THIN CLIENT MAYBE?
On Apr 12, 2005 9:38 AM, Chuck Bunn [EMAIL PROTECTED] wrote: Hi, Actually I guess what I am looking for is semi-sealed box that I can add 1 or 2 PCI cards too. A regular PC work work in most cases since I do not want a keyboard or mouse attached to it. I do not want users screwing with the system. If it is sealed with no monitor/keyboard/mouse then they can't screw it up very easily. I guess I am looking for something that is somewhere in between a PC and Linksys router box. One possibility might be a thin client box, but I haven't found any sources for an OEM box. I looked at the HP (http://h18004.www1.hp.com/products/thinclients/index_t5000.html) thin clients but I can get a Dell Box for the same price that does more. Thanks Have you checked eBay? http://lists.digium.com/pipermail/asterisk-users/2005-April/100861.html ___ 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] QoS TOS numbers and Cisco IOS
Does anyone know how setting the TOS bits in iax.conf corresponds to the Cisco TOS types? For example, if I set: tos=0x04 in iax.conf, and on the Cisco, I use: access-list 110 permit ip any any tos 4 I can't get the Cisco to match any packets. I've tried various combinations of numbers on both asterisk and the cisco. I've also tried hex to decimal conversion. I just can't get the Cisco to see the TOS bits that I set in iax.conf. Thanks, Noah ___ 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] Noises on ZAP Channels
Hi everyone, I have the following annoying problem with my Digium TE410 Quad-Pri-Card: I sometimes hear strange noises on bridged calls from our PBX to the PSTN (a colleage called it clipping?) We have the following setup running: PSTN - Asterisk - od PBX (Trunk one to the PSTN, Trunk two to our existing PBX) Currently our System does not really do quite a lot, since almost every call is currently terminated on the old PBX. Here comes my zaptel.conf: span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 loadzone=nl defaultzone=nl And my zapata.conf (comments removed) [channels] faxdetect=none language=de context=default switchtype=euroisdn pridialplan=unknown overlapdial=yes signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes restrictcid=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=no echocancelwhenbridged=no (we turned the echo-canceling of, because we had problems while sending/receiving faxes) echotraining=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no musiconhold=default signalling = pri_cpe group = 1 pridialplan=unknown musiconhold=default context = incoming channel = 1-15,17-31 signalling = pri_net group = 2 pridialplan=local musiconhold=default context = from_pbx channel = 32-46,48-62 In my extensions.conf i use some PHP-AGI Skripts for dialing and call-processing. I hope someone can give me a hint what to do/improve or what to search for. So far i did not find anything useful on the net Thanks in advance, Carsten Bock ___ 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] Agents
I am a little confused as to the purpose of agents. My old phone system required that a user/agent be logged into a phone in order to use that phone, regardless if the agent was joining a Queue. It seems that agents in the context of Asterisk are more for dealing with Queues. So it seems that if I am not using any Queues then there is no reason to you have agents. I suppose the sip registry is really the equivilant of the old Login/Logout routine in my old system. Is this correct? ___ 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] Multiple Servers and 1 Central Voicemail
Also, what happens if for example, the user is accessing his VMB on server 1 and changes his password, then travel to where server 2 is and tries to access his VMB? the config on server2 would still have the old one so you need to sync voicemail.conf on all servers too ... If you use the realtime config via a DB, it should be OK. But I still don't think that MWI will work properly if a message is left on server A and user is actually registered on server B, which is NOT on the same network and hence does NOT share the same voice mail spool. How will B know there is a message left on A for the same user? Does realtime share this info too? And if so, how does the message get retrieved if B does not have access to files on server A, where the actual message is? --Luki ___ 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] multiple line usage on Polycom IP300
On Tuesday 12 April 2005 11:00 am, MobilPete wrote: we tried both, setting it as same and also seperate. but niether worked. I've never used the IP300, but I do have an IP500 on our network. It has 3 line buttons, each line can do 2 simultaneous calls. Each line button registers as its own SIP device (op-1, op-2, and op-3). I wrote an AGI script to dial the IP500. It uses the * Manager to do 'show channels' to find the line button on the IP500 with the least number of simultaneous calls (e.g. which SIP device [SIP/op-1, SIP/op-2, or SIP/op-3]) then the AGI script just redirects the call to the next available line on the IP500 using AGI 'EXEC' to run the 'Dial' app. If anybody is interested in the script, ill try to clean it up enough to post. -josiah - Original Message - From: Josiah Bryan [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 12, 2005 9:41 AM Subject: Re: [Asterisk-Users] multiple line usage on Polycom IP300 On Tuesday 12 April 2005 10:18 am, MobilPete wrote: can anyone help ?? trying to get Polycom IP300 to utilize both lines, would like calls to roll to open line when incoming call arrives while user is on line 1. Looked everywhere and tried many things with no luck. Do you have your lines register sepratly? E.g. is there a seperate entry in sip.conf for each line or do they both register as the same sip device? -- 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 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: polycom phones
On Apr 11, 2005 11:49 PM, Greg Boehnlein [EMAIL PROTECTED] wrote: On Mon, 11 Apr 2005, Noah Miller wrote: This this may sound ridiculous, but we've had problems with this when the users did not plug the handset cord in completely. 8 out of our 12 employees made the mistake, as the plug on the IPX00's appears to be all the way in when it is actually not. Not ridiculous at all. We had the same problem. In fact, the cord will click into place when it's not really all the way in. I had the same problem.. :) aolMe too!/aol Took a few minutes to figure it out... was sweating bullets. ___ 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] Multiple TDM cards on the same box
Hi All, I'm trying to install 2 TDM400x cards on the same [EMAIL PROTECTED] box, and I've currentlyhaving issues where one card is identified by ztfcg, and the other isn't at all. Any idea what i may be doing wrong here? has anyone got an [EMAIL PROTECTED] working in such a manner? Nir S ___ 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 I reduce echo on the Caller side
Hi, I get an echo only from the caller end when I am making calls. I only get it for some VOIP providers. I am using asterisk Asterisk CVS-v1-0-03/26/05-16:54:47 and Grandstream HandyTone 486 and 488. My default codec is ulaw. Is there any way I can reduce the echo without comprising quality? Thanks Joel ___ 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] Low cost box for hosting Asterisk and atleastone TDM400p - THIN CLIENT MAYBE?
What I use. At provantage.com Part Description Price ANTG02V Antec Mini-Tower with 8 Drive Bays - BLACK 45.93 ASUS1FQ ASUS A7V400-MX Motherboard KM400A 400/333FSB VID LAN 3PCI 49.49 AAMD16U AMD Sempron 2600+ Processor-In-A-Box 77.54 SEGE155 Seagate Barracuda 7200.7 40GB EIDE ATA-100 7200 RPM 3.5LP FDB 49.46 KINM13T Kingston 256MB 400MHz DDR PC3200 DIMM 3-3-3 25.36 Subtotal: 247.78 Shipping: 42.25 Total Order:$ 290.03 You can double the ram for about 30 bucks and add a DVDrom for about $35 with shipping included. On Apr 12, 2005 8:38 AM, Chuck Bunn [EMAIL PROTECTED] wrote: Hi, Actually I guess what I am looking for is semi-sealed box that I can add 1 or 2 PCI cards too. A regular PC work work in most cases since I do not want a keyboard or mouse attached to it. I do not want users screwing with the system. If it is sealed with no monitor/keyboard/mouse then they can't screw it up very easily. I guess I am looking for something that is somewhere in between a PC and Linksys router box. One possibility might be a thin client box, but I haven't found any sources for an OEM box. I looked at the HP (http://h18004.www1.hp.com/products/thinclients/index_t5000.html) thin clients but I can get a Dell Box for the same price that does more. 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 -- Andrew Latham http://www.lathama.com [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] If any of the above are not working, we have bigger problems than my email. ___ 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] LCDial and default provider
Does anybody know how I could set a default provider for LCDial? Also, how could I use it for national calls, dialling without international prefix? TIA, Alex ___ 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] Dumb question ?
Here it is
exten = s,1,answer
exten = s,2,SetCIDName('PMG')
In a lot of config files I see exten = s,snip ..
Is s just an extension or system variable for all extensions ? or
something else ?
Thanks
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Re: [Asterisk-Users] QoS TOS numbers and Cisco IOS
Does anyone know how setting the TOS bits in iax.conf corresponds to the Cisco TOS types? For example, if I set: tos=0x04 in iax.conf, and on the Cisco, I use: access-list 110 permit ip any any tos 4 I can't get the Cisco to match any packets. I've tried various combinations of numbers on both asterisk and the cisco. I've also tried hex to decimal conversion. I just can't get the Cisco to see the TOS bits that I set in iax.conf. Here's what I'm using. sip.conf: tos=0x18 ;lowdelay ;sets ip tos bits (=lowdelay, throughput) iax.conf: tos=lowdelay Cisco: class-map match-all voice-rtp match access-group 103 access-list 103 permit ip any any tos min-delay access-list 103 permit ip any any tos 12 C1750#show access-list 103 Extended IP access list 103 permit ip any any tos min-delay (2077271 matches) permit ip any any tos 12 (651833 matches) The NAI Sniffer does a better job of showing the bits. Here's two samples for the above: sip packet (tos=0x18): IP: Type of service = 18 IP: 000. = routine IP: ...1 = low delay IP: 1... = high throughput IP: .0.. = normal reliability IP: ..0. = ECT bit - transport protocol will ignore the CE bit IP: ...0 = CE bit - no congestion iax packet (tos=lowdelay): IP: Type of service = 10 IP: 000. = routine IP: ...1 = low delay IP: 0... = normal throughput IP: .0.. = normal reliability IP: ..0. = ECT bit - transport protocol will ignore the CE bit IP: ...0 = CE bit - no congestion Study the above and the bits become very clear. :) ___ 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] Multiple Servers and 1 Central Voicemail
On Tuesday 12 April 2005 11:49 am, Luki wrote: Also, what happens if for example, the user is accessing his VMB on server 1 and changes his password, then travel to where server 2 is and tries to access his VMB? the config on server2 would still have the old one so you need to sync voicemail.conf on all servers too ... If you use the realtime config via a DB, it should be OK. But I still don't think that MWI will work properly if a message is left on server A and user is actually registered on server B, which is NOT on the same network and hence does NOT share the same voice mail spool. How will B know there is a message left on A for the same user? Does realtime share this info too? And if so, how does the message get retrieved if B does not have access to files on server A, where the actual message is? Why not just NFS mount the /var/spool/asterisk/voicemail directory from a central server? That way, all servers share the same spool and the MWI will get reflected on all servers. -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
Re: [Asterisk-Users] DS3000P - 20 E1 capacity on single card
Bicom Systems wrote: What is target release date for DS3000P? That has not been announced; sometime after today would be a safe assumption :-) ___ 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
