Re: [Asterisk-Users] Asterisk Hardware recomendation
Yes, transcoding is not going to work for that density. asterisk doesn't do g723, and even if it would your system would not be able to handle more than 150 simultaneous g711 to g729/g723 transcodings. If you would go for plain g711, you could do 500, but i don't recommend it, especially if you have little asterisk experience. (i'd say go for a cluster). Zoa www.asteriskguru.com Krystian Filiks wrote: I will be using IP Hard and soft phones all the way, so everything will be on Ethernet, for this I want 1Gbit incoming and 1Gigabit outgoing, looking for atleast 500 simultaneous calls, with 2 3.6Ghz processors I think I could squize out more then that. For codec I want to use g711 on the outgoing as it will only be over local lan and just about 2 meter away from the termination point (so almost 0 in loss) as for incoming I think g.729 or 723 maybe GSM. I know that the recoding take the most of the CPU power so perhaps I can do g.7xx codec all the way, that is a mather of test and see. No other cards in the box then LAN cards. On top of that I'll run voicemail, text to speech and music on hold. Any comments? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: den 8 december 2005 02:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Hardware recomendation Krystian - what kind of port density are you aiming for? Will you be running analog or digital? Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Krystian Filiks wrote: Hello asterisk people! I have been running a test * server a P III box for some time now and it's been rock stable. Now I'm looking to build a production system with as big capacity as possible on 2 Xeon 3.6Ghz processors. I'm wondering what you are thinking about Supermicro 6014H-32 SuperServer with Dual 3.6Ghz Xeon processors and 2M casche each, 2 X Gigabit LAN ports, 1Gb of RAM and about 80Gb of SATA HDD. For the OS I was thinking about Debian and the latest stable release of Asterisk. I will be using IP to IP technology without any PRI cards only IP to IP. Clients will be using SIP and Aserisk will terminate on to H.323 or possibly SIP How can I benchmark this thing (Aprox) without having to buy the server? Has any one had any experience of such server? Please comment. --- - ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HOW TO: CDR Customer IP address where call came in from
Rehan Ahmed ha scritto: I dont see the ip in the Master.csv but you can view the IP when the call comes in on the CLI Window. I am guessing there must be a command or a way to record this ip in your CDR using AGI, we are using agi to make our own CDR but i would apreciate if some one can tell how to record the IP address of the caller. If you install the iaxusers/sipusers mysql backend (which everyone seems to call realtime) the ip will be stored in a 'ipaddr' column. You can put a select in some agi to retrieve the IP of the peer. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Hardware recomendation
What about plain g729? My main concern is the Hardware, anyone that can tell me if this Supermicro 6014H-32 is stable and sutible for asterisk? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoa Sent: den 8 december 2005 09:00 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Hardware recomendation Yes, transcoding is not going to work for that density. asterisk doesn't do g723, and even if it would your system would not be able to handle more than 150 simultaneous g711 to g729/g723 transcodings. If you would go for plain g711, you could do 500, but i don't recommend it, especially if you have little asterisk experience. (i'd say go for a cluster). Zoa www.asteriskguru.com Krystian Filiks wrote: I will be using IP Hard and soft phones all the way, so everything will be on Ethernet, for this I want 1Gbit incoming and 1Gigabit outgoing, looking for atleast 500 simultaneous calls, with 2 3.6Ghz processors I think I could squize out more then that. For codec I want to use g711 on the outgoing as it will only be over local lan and just about 2 meter away from the termination point (so almost 0 in loss) as for incoming I think g.729 or 723 maybe GSM. I know that the recoding take the most of the CPU power so perhaps I can do g.7xx codec all the way, that is a mather of test and see. No other cards in the box then LAN cards. On top of that I'll run voicemail, text to speech and music on hold. Any comments? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: den 8 december 2005 02:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Hardware recomendation Krystian - what kind of port density are you aiming for? Will you be running analog or digital? Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Krystian Filiks wrote: Hello asterisk people! I have been running a test * server a P III box for some time now and it's been rock stable. Now I'm looking to build a production system with as big capacity as possible on 2 Xeon 3.6Ghz processors. I'm wondering what you are thinking about Supermicro 6014H-32 SuperServer with Dual 3.6Ghz Xeon processors and 2M casche each, 2 X Gigabit LAN ports, 1Gb of RAM and about 80Gb of SATA HDD. For the OS I was thinking about Debian and the latest stable release of Asterisk. I will be using IP to IP technology without any PRI cards only IP to IP. Clients will be using SIP and Aserisk will terminate on to H.323 or possibly SIP How can I benchmark this thing (Aprox) without having to buy the server? Has any one had any experience of such server? Please comment. -- - - ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Hardware recomendation
How would one go about to implement such a cluster? How do the different Asterisk boxes know of the extensions on all the other boxes? Is each client bound to it's box or can it connect to any box in the cluster, ie if one fails can the other take over and share the load of the failed on between themselves? I would be very interested in hearing more of such solutions and people experiences with it. Regards, Kristian Larsson On Thu, Dec 08, 2005 at 10:00:01AM +0200, Zoa wrote: Yes, transcoding is not going to work for that density. asterisk doesn't do g723, and even if it would your system would not be able to handle more than 150 simultaneous g711 to g729/g723 transcodings. If you would go for plain g711, you could do 500, but i don't recommend it, especially if you have little asterisk experience. (i'd say go for a cluster). Zoa www.asteriskguru.com Krystian Filiks wrote: I will be using IP Hard and soft phones all the way, so everything will be on Ethernet, for this I want 1Gbit incoming and 1Gigabit outgoing, looking for atleast 500 simultaneous calls, with 2 3.6Ghz processors I think I could squize out more then that. For codec I want to use g711 on the outgoing as it will only be over local lan and just about 2 meter away from the termination point (so almost 0 in loss) as for incoming I think g.729 or 723 maybe GSM. I know that the recoding take the most of the CPU power so perhaps I can do g.7xx codec all the way, that is a mather of test and see. No other cards in the box then LAN cards. On top of that I'll run voicemail, text to speech and music on hold. Any comments? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: den 8 december 2005 02:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Hardware recomendation Krystian - what kind of port density are you aiming for? Will you be running analog or digital? Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Krystian Filiks wrote: Hello asterisk people! I have been running a test * server a P III box for some time now and it's been rock stable. Now I'm looking to build a production system with as big capacity as possible on 2 Xeon 3.6Ghz processors. I'm wondering what you are thinking about Supermicro 6014H-32 SuperServer with Dual 3.6Ghz Xeon processors and 2M casche each, 2 X Gigabit LAN ports, 1Gb of RAM and about 80Gb of SATA HDD. For the OS I was thinking about Debian and the latest stable release of Asterisk. I will be using IP to IP technology without any PRI cards only IP to IP. Clients will be using SIP and Aserisk will terminate on to H.323 or possibly SIP How can I benchmark this thing (Aprox) without having to buy the server? Has any one had any experience of such server? Please comment. --- - ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Hardware recomendation
Krystian Filiks wrote: What about plain g729? My main concern is the Hardware, anyone that can tell me if this Supermicro 6014H-32 is stable and sutible for asterisk? Supermicro Superservers are traditionally extremely stable and reliable. -- Roman Volf Keystreams Internet Solutions [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip behind the NAT
can u paste your sip.conf general section,,? there have another possible cause... the both side use different codecm and asterisk can not translaste it ... -- Jeffery On 12/8/05, chawki hammoud [EMAIL PROTECTED] wrote: Hi:i added these two lines to my general context ,butnothing happened the same result the sound came in one way for 3 seconds and stopped but it didnt hangup.--- Jeffery Chen [EMAIL PROTECTED] wrote: If your Astersik server behind NAT too, your need modify SIP.conf like this externalIP= x.x.x.x localnet= x.x.x. hope this can help you On 12/8/05, Moises Silva [EMAIL PROTECTED] wrote: what type of NAT do you have? sync? full cone? cone restricted, port restricted? any messages in asterisk verbose console? best regards On 12/7/05, chawki hammoud [EMAIL PROTECTED] wrote: Hi list: i have an asterisk box behind the NAT ,when i try to send calls through Sip to the voip provider server the call is answered but in a one way calling,I hear the voice of the other side just for 4 seconds and then stop but the call do not hangup. my sip.conf is: [voip provider] type=peer host=213.112.50.8 username=XXX secret=XX fromuser=XXX canreinvite=no nat=yes insercure=invite disallow=all allow=gsm __ Yahoo! DSL – Something to write home about. Just $16.99/mo. or less. dsl.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.comhttp://easynews.com/-- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jeffery ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam?Yahoo! Mail has the best spam protection aroundhttp://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo cancellation
I am having problems with echo, first let me explain my setup: I have a Gateway box, which basically is an Asterisk with a PRI card. It's only job is to interface with 2 incoming ISDN PRI connections. Then there are two other asterisk boxes to which my users are registered. Dialing from a phone it hits the first asterisk which forwards it to the gateway box and then on to the PSTN. What are the general causes of echo? When calling from my SIP phone I hear no echo but the other end, the PSTN end, hears a lot of echo. What could cause this? Kristian. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE:how to listen voicemail messages
Take a look at the function VoiceMailMain() You will need to put it somewhere in your extensions.conf. On 12/8/05, Tejas Shah [EMAIL PROTECTED] wrote: hi all, I have made two voice mail boxes for 2 users on asterisk server (/var/spool/asterisk/voicemail/testmail/inside this 2 boxes for 2 users). i have made following settings in voicemail.conf : [testmail] vipul=,vipul patel, [EMAIL PROTECTED] tejas=,tejas shah,[EMAIL PROTECTED] i have made appprpriate settings in SIP.CONF and EXTENSIONS.CONF. now when any of the user is unavailable voicemail is getting active. It is also allowing to record messages on voicemail box. now my problem is.suppose for one user say TEJAS another user has send voicemail. then how that user TEJAS can listen voicemail message. Is there any command to run on asterisk server. how can i access my voicemail? thanks tejas Yahoo! Shopping Find Great Deals on Holiday Gifts at Yahoo! Shopping ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tijmen van den BrinkWilhelminaweg 463441 XC WoerdenTel: 0642233831MSN: [EMAIL PROTECTED]Skype: [EMAIL PROTECTED]SIP:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on PPC chan_capi issue
Hello Patrick, I have an Eicon Diva PRI-30M card and use the Eicon Linux drivers with chan_capi_cm. I am able to do ISDN to SIP calls with this. Have you tried using the Eicon drivers instead, rather than zaptel and zib pri. Instruction for doing this can be found here: http://www.voip-info.org/wiki/view/Asterisk+Eicon+Diva+CAPI+ISDN I hope this helps Thanks David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Patrick Sent: 06 December 2005 01:40 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk on PPC chan_capi issue Hi all, I have a PPC box (IBM RS6000 43P-150, bigendian afaik) which runs Fedora Core 5 Test1 and zaptel, libpri and asterisk 1.2.0. I also installed chan_capi (0.6.1) so I can use my Eicon Diva Server BRI card. Asterisk was compiled with DEBUG=-g and DEBUG_THREADS = -DDUMP_SCHEDULER -DDEBUG_SCHEDULER-DDEBUG_THREADS -DDO_CRASH -DDETECT_DEADLOCKS. Next I did make clean, make valgrind, make install. Asterisk runs as user/group asterisk/asterisk. SIP -- SIP calls are fine, Calls from SIP out to the PSTN via CAPI/ISDN are fine too. ISDN/CAPI -- SIP calls don't work. Example output of the issue is below. Anyone have an idea how I fix this? Thanks and regards, Patrick chan_capi registers fine: ** [chan_capi.so] = (Common ISDN API for Asterisk) == This box has 1 capi controller(s). == Reading config for BRI1 -- ast_capi_pvt BRI1-pseudo-D (MSN1,MSN2,capi-in,0,2) (1,4,128) -- ast_capi_pvt BRI1 (MSN1,MSN2,capi-in,0,2) (1,4,128) -- ast_capi_pvt BRI1 (MSN1,MSN2,capi-in,0,2) (1,4,128) -- listening on contr1 CIPmask = 0x1fff03ff == Registered channel type 'CAPI' (Common ISDN API Driver ($Revision: 1.115 $) ) == Registered application 'capiCommand' == Registered custom function VANITYNUMBER Call from my GSM to a SIP phone (exten 1003) via ISDN/CAPI (MSN2): ** == BRI1: Incoming call 'my GSM' - 'MSN2' -- Executing Macro(CAPI/BRI1/MSN2-0, stdexten|1003|SIP/1003) in new stack -- Executing Dial(CAPI/BRI1/MSN2-0, SIP/1003|10|TtwW) in new stack Dec 6 02:30:47 WARNING[28889]: channel.c:2494 ast_request: No translator path exists for channel type SIP (native 65535) to 0 Dec 6 02:30:47 NOTICE[28889]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing Goto(CAPI/BRI1/MSN2-0, s-CHANUNAVAIL|1) in new stack -- Goto (macro-stdexten,s-CHANUNAVAIL,1) -- Executing Goto(CAPI/BRI1/MSN2-0, s-NOANSWER|1) in new stack -- Goto (macro-stdexten,s-NOANSWER,1) -- Executing Answer(CAPI/BRI1/MSN2-0, ) in new stack == BRI1: Answering for 703241494 -- Executing Wait(CAPI/BRI1/MSN2-0, 1) in new stack Dec 6 02:30:47 NOTICE[28889]: channel.c:1893 ast_read: Dropping incompatible voice frame on CAPI/BRI1/MSN2-0 of format alaw since our native format has changed to unknown Dec 6 02:30:47 NOTICE[28889]: channel.c:1893 ast_read: Dropping incompatible voice frame on CAPI/BRI1/MSN2-0 of format alaw since our native format has changed to unknown [snipped tons more of these] Dec 6 02:30:48 NOTICE[28889]: channel.c:1893 ast_read: Dropping incompatible voice frame on CAPI/BRI1/MSN2-0 of format alaw since our native format has changed to unknown -- Executing VoiceMail(CAPI/BRI1/MSN2, u1003) in new stack Dec 6 02:30:48 WARNING[28889]: channel.c:2313 set_format: Unable to find a codec translation path from unknown to gsm Dec 6 02:30:48 WARNING[28889]: file.c:820 ast_streamfile: Unable to open vm-theperson (format unknown): No such file or directory == BRI1: CAPI Hangingup CAPI INFO 0x3490: Normal call clearing ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Integration of external (ZAP) agents into queue
Hey, I´m just wondering if its possible to login an ISDN phone connected to my asterisk box via an S0 Trunk line. My setup is: Siemens Hipath 4000 S0 Asterisk --- SIP --- x SIP Phones I would like to setup several queues on my asterisk and allow both SIP Users as well as the external ZAP devices to register as agents in the queue. At the moment my queue is working, a Siemens user is able to dial the login extension which looks like the following: -- Accepting voice call from '3331' to '3299' on channel 0/1, span 4 -- Executing Macro(Zap/10-1, agent-add|101|101) in new stack -- Executing Wait(Zap/10-1, 1) in new stack -- Accepting voice call from '3331' to '3299' on channel 0/1, span 4 -- Executing Macro(Zap/10-1, agent-add|101|101) in new stack -- Executing Wait(Zap/10-1, 1) in new stack -- Executing GotoIf(Zap/10-1, 0?4:3)) in new stack -- Goto (macro-agent-add,s,3) -- Executing Authenticate(Zap/10-1, 101) in new stack -- Playing 'agent-pass' (language 'de') -- Executing GotoIf(Zap/10-1, 0?4:3)) in new stack -- Goto (macro-agent-add,s,3) -- Executing Authenticate(Zap/10-1, 101) in new stack -- Playing 'agent-pass' (language 'de') -- Playing 'auth-thankyou' (language 'de') -- Playing 'auth-thankyou' (language 'de') -- Executing AddQueueMember(Zap/10-1, 101) in new stack -- Executing AddQueueMember(Zap/10-1, 101) in new stack Dec 8 11:44:48 NOTICE[10197]: app_queue.c:2812 aqm_exec: Added interface 'Zap/10' to queue '101' Added interface 'Zap/10' to queue '101' -- Executing Wait(Zap/10-1, 1) in new stack -- Executing Wait(Zap/10-1, 1) in new stack -- Executing Playback(Zap/10-1, agent-loginok) in new stack -- Playing 'agent-loginok' (language 'de') -- Executing Playback(Zap/10-1, agent-loginok) in new stack -- Playing 'agent-loginok' (language 'de') -- Executing Hangup(Zap/10-1, ) in new stack == Spawn extension (macro-agent-add, s, 7) exited non-zero on 'Zap/10-1' in macro 'agent-add' == Spawn extension (isdn-incoming, 3299, 1) exited non-zero on 'Zap/10-1' -- Executing Hangup(Zap/10-1, ) in new stack == Spawn extension (macro-agent-add, s, 7) exited non-zero on 'Zap/10-1' in macro 'agent-add' == Spawn extension (isdn-incoming, 3299, 1) exited non-zero on 'Zap/10-1' -- Hungup 'Zap/10-1' -- Hungup 'Zap/10-1' When I do: bit144*CLI show queue 101 101 has 0 calls (max 2) in 'fewestcalls' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: Zap/10 (dynamic) (Not in use) has taken no calls yet No Callers The ZAP endpoint seems to be registered in the queue but when I call my queue, the call is not forwarded to the right (external extension): -- Executing Answer(SIP/6002-30fd, ) in new stack -- Executing SetCIDName(SIP/6002-30fd, Hotline - Snom 190) in new stack -- Executing Queue(SIP/6002-30fd, 101|t|||120) in new stack -- Started music on hold, class 'default', on SIP/6002-30fd -- Requested transfer capability: 0x00 - SPEECH -- Called Zap/11 -- Executing Answer(SIP/6002-30fd, ) in new stack -- Executing SetCIDName(SIP/6002-30fd, Hotline - Snom 190) in new stack -- Executing Queue(SIP/6002-30fd, 101|t|||120) in new stack -- Started music on hold, class 'default', on SIP/6002-30fd -- Requested transfer capability: 0x00 - SPEECH -- Called Zap/11 -- Nobody picked up in 1 ms -- Hungup 'Zap/11-1' -- Nobody picked up in 1 ms -- Hungup 'Zap/11-1' I seems like only the port (in this case Zap/11-1) is registered but not the numbered extension. So when a call is to be forwarded by the queue it dows not know where to direct it. Is it possible to do this anyhow? Or did anybody realize something like this before? Thanks in advance Michael ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as sipclient
Hi! How many register lines does Asterisk support i sip.conf? I may need a setup where Asterisk should act as a many sipclient (over 10.000) (seen from the telco side) and then forward the calls to the real sipclients. Is that possible? Is it possible to use SER to do this instead of Asterisk?-- Morten Isaksenhttp://www.misak.dk/blog/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SVN Revision 7230
hello, I always update trough CVS from the cvs tree but i only see this revision 7230 in the asterisk all the days but the changelog say there are already newer versions. Did i updated wrong or is the revison wrong? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A2billing signalling
I have setup a2billing to be used as more of a billing software. I have made a card for each of my customers and the call through and get authenticated with callerID and then the call is placed using dnid. This is working great. The problem that I just ralized is that when I get a call asterisk answers the call to send it to a2billing. This process is signaling to the calling party that the call has been answered already. I am trying to get a2 billing to take the call but only signal to the calling party that the call has been answered when the call has actually started. I am guessing this would need modification done to the a2billing.php file in the agi-bin folder. I dont know anything about agi. Can anyone please help on this issue. Thanks in advance Zafer ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MYSQL cmd with Asterisk Realtime
Hi all,
I have problem to get MYSQL cmd work with Asterisk Realtime (Asterisk
1.2.1 and Asterisk-addon 1.2.1). It happened like what being described
in this post
http://lists.digium.com/pipermail/asterisk-users/2005-May/107956.html.
The ${resultid}reference on the fetch line using asterisk realtime have
no value.
Is this a bug or my wrong configuration?
Please advice.
Regard
Akhng
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[Asterisk-Users] dtmf problem
Hi, I got this message in the asterisk console while sending the dtmf from a phone.Dec 8 14:55:50 WARNING[29098]: codec_ilbc.c:163 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? Please help me to solve this.Thanks Jibu Yahoo! India Matrimony: Find your partner now.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wrong caller id num on Swissvoice IP10S
Hello, I'm using Swissvoice IP10S phones. When I dial from number 11 to number 22, on the phone with number 22 there is displayed a name set in sip.conf for user 11 and a number 22 (not 11). When I try to call back, it calls to the 22 (self). How to correct this? In sip.conf I have: callerid=Someone 11 I've even tried this in extensions.conf: exten = 22,1,Set(CALLERID(num)=11) exten = 22,2,Dial(SIP/22) But it doesn't work. NoOp says the correct CALLERIDNUM. What's wrong? -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX
Russ Price wrote: So, are there any IP faxes? Sort of. But I'm talking about hardware IP faxes. -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk with EWSD v16
Dear Gulzar, Thank you for your reply, I am using same configs. I have tried both 0 1 in timing but no luck. I will try again with 'timing' parameter = 1 in zapata.conf best Regards, -- Atif Rasheed Gulzar Hussain wrote: I am using EWSD's PRIs and I am not having this problem my configs are Zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone = us defaultzone=us Zapata.conf [channels] language=en context=ext-acd switchtype=euroisdn signalling=pri_cpe echocancel=yes echocancelwhenbridged=yes group=1 channel = 1-15 channel = 17-31 pridialplan=private prilocaldialplan=private overlapdial=yes usecallerid=yes hidecallerid=no immediate=no usecallingpres=no --- Atif Rasheed [EMAIL PROTECTED] wrote: if any EWSD guru out there..please help ??? Hello all, I am running Asterisk with Digium E1 card with zaptel, libpri, asterisk cvs v1-2. My server is interfaced with EWSD v16 using a PRI on E1. I am running into a problem that at my telco's end alot of trunks are getting BPRM (Block permanant) status. I am not sure why EWSD is blocking trunks. config at my end::: coding = hdb3 format = ccs,crc4 signalling = euroisdn, pri_cpe config at my telco's end coding = hdb3 format = crc4mf signalling = euroisdn, pri_net Is there any EWSD guru around who can explain why trunks are getting BPRM status in EWSD switch. I will really appriciate your help Thank you -- Atif Rasheed ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR manipulation in macros
Hi all,
I'm trying to change the CDR userfield in a macro which is executed upon
call pickup (option M in Dial command). The goal is to log the answer
time (in the default CDR it is not correct as the call is picked up to
play music on hold to the caller before Dialing the called extension). I
use Asterisk 1.0.9, with asterisk-oh323 0.6.5.
Here is my dialplan:
...
exten = s,8,Dial(OH323/[EMAIL PROTECTED]:1720,20,mM(CdrAnswerDate)) ;
execute macro-CdrAnswerDate when the called extension 1234 is picked up
exten = s,9,AppendCDRUserField(no_answer ) ; if no answer after 20
sec.
...
The macro-CdrAnswerDate is defined as follows:
[macro-CdrAnswerDate]
exten = s,1,AGI(getCurrentTimeDate.sh) ; shell script that sets the
variable ANSWER_DATE to the pickup date
exten = s,2,AppendCDRUserField(answered ${ANSWER_DATE})
Here is what I get in the console:
-- Started music on hold, class 'default', on
OH323/HiPath4000,@10.253.3.27-393a
H.323 call 'ip$localhost/18575', exception CALL_ALERTED.
-- OH323/[EMAIL PROTECTED] is ringing
H.323 call 'ip$localhost/18575', exception CALL_ESTABLISHED.
-- OH323/[EMAIL PROTECTED] answered
OH323/HiPath4000,@10.253.3.27-393a
-- Executing AGI(OH323/[EMAIL PROTECTED],
getCurrentTimeDate.sh) in new stack
-- Launched AGI Script /usr/local/asterisk/agi/getCurrentTimeDate.sh
getCurrentTimeDate.sh: Call answered 2005-12-08 11:04:51
-- AGI Script getCurrentTimeDate.sh completed, returning 0
-- Executing AppendCDRUserField(OH323/[EMAIL PROTECTED],
answered 2005-12-08 11:04:51) in new stack
-- Stopped music on hold on OH323/HiPath4000,@10.253.3.27-393a
-- H.323 call 'ip$localhost/18575' cleared, reason 4 (Cleared by
remote user), established (2 sec)
-- Hungup 'OH323/[EMAIL PROTECTED]'
Which seems to indicate that it is OK.
However the recorded CDR Userfield (I use MySQL for that) is not
updated: it contains only the values I had Append-ed before...
Is there a problem with changing CDRs in macros? My previous tests
showed that using ForkCDR or ResetCDR in macros doesn't work either.
Theank you for your help.
Regards,
Silviu
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RE: [Asterisk-Users] Streaming MOH
Title: Message
Hi
all,
Did
any one succeeded to configure MOH (Asterisk 1.2.0) to play audio from the
streaming source?
Sample
musiconhold.conf has entry like this:
[stream]
mode=custom
application=/usr/sbin/streamplayer192.168.100.132
8088 format=ulaw
I used
JMStudio (java JMF application) to transmit the audio.
What
other app can I use to create audio stream?
Asterisk console shows only:
-- Executing MusicOnHold("SIP/213.240.56.20-091bfe60",
"stream") in new stack -- Started music on hold, class
'stream', on channel 'SIP/213.240.56.20-091bfe60'
...but
there is no music.
Regards,Stojan
Sljivic
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stojan
Sljivic - GDSSent: Tuesday, December 06, 2005 17:59To:
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject:
RE: [Asterisk-Users] Streaming MOH
Hi,
What
application can I use to stream the audio for "streaming audio
MOH"?
Regards,Stojan
Sljivic
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stojan
Sljivic - GDSSent: Monday, December 05, 2005 14:24To:
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject:
[Asterisk-Users] Streaming MOH
Hi,
Have someone successfully configured the streaming MOH in Asterisk
1.2.0 using streamplayer?
Regards
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[Asterisk-Users] GROUP_COUNT and AGI
hi, is it possible to use GROUP_COUNT function in AGIs. i could not make it work. :-( thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Change Inbound CALL ID Asterisk
Hey everybody, Is here anyway to change the name asterisk on the caller id inbound to the client/sip app? Thanks, Oliver ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Change Inbound CALL ID Asterisk
Oliver Vermeulen wrote: Is here anyway to change the name asterisk on the caller id inbound to the client/sip app? In sip.conf type in the section of desired user: callerid=Caller Name 11 Where 11 is your caller number. -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX stress test
Hi, Is there any existing tools for IAX stress testing? I need to know with how mach IAX - H.323 and IAX - IAX simultaneous calls can my server works. Now it is few clients, but in the future it will be mach more. Can I emulate many dummy IAX clients on single computer? -- Thanks, Eugene Prokopiev ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No application 'MeetMe' for extension
Hi all! I have an [EMAIL PROTECTED] 2.1 setup here which is working 99% the way it should. The only thing that does not work is Meetme/Conferences... In the log-file I see: Dec 8 11:51:28 WARNING[3288] pbx.c: No application 'MeetMe' for extension (from-internal, 8125, 6) This is when I dial 8125 from extension 125. 8125 is defined in the meetme(-additional).conf And before you ask: yes, ztdummy is loaded... Who has any suggestions? I'm stumped... :-/ Regards, Evert ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] multiple line registrations on attendant console
Hi all, I've noticed that Polycom and Snom each offer attendant console expansions. As far as I understand, the point in using all thebuttons they provide is to program them to register as extensions in order to be able to monitor the status of each extension at any given time and, also, to be able to pickup any extension that has been ringing for a while. Is this feature supported by asterisk? I have tried to register two sip phones with the same extension but when I dial the extension only one of the phones rings. This is a feature that is necessary in many telephony environments and I'd be happy to be able to implement it using asterisk. I've searched through the list's postsfor the past few months but didn't come up with anything usefull. any help would be greatly appreciated Dionisis ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] about * and CM/CME
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, on voip-info there's two howtos to connect asterisk with CM and/or CME Cisco, but always with sip trunk. What about h323 instead of sip? there's someone that has tested something like that? MWI will work too? Your feedbacks will be appreciated Best Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFDmBXQMakHrsrHP9wRAmkCAJ93qnK23Ylkdm+qbALMQByjysYAPwCgpp+U wuuJljz0FEDoFUhTqAfBNpI= =nEti -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] about g729
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, my topology is like that: ISP --[sip]-- Asterisk --[sip]-- CME Cisco -- ip phones ISP services are g711 and g729 enabled. My Asterisk is registered on ISP with two sip UA. Then I've forwarded calls from ISP to ip phones registered on CME Cisco. With g711 all works like a charm, but for audio quality, and bandwidth utilization, I'm trying now to work with g729 between CME and ISP. What about Asterisk? this is a pass-thru example, or maybe I've to pay a g729 license? Thanks for your support Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFDmBioMakHrsrHP9wRAr2+AJkBgIkiBa6R2mayleAdDM8U505b9wCeIdmu 61KS6xIesH47QknyZI04Gy4= =STDC -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Ringtone when dialing
yusuf wrote: Hi all, can anyone tell me when (or how) * starts generating ring tone when a call is made. The reason I ask this is I have an E1 coming from a PBX into my * box (CVS 19/07/2005). I have some intermittent problems. 1. Sometimes no ringtone is generated, so I dial a number, and the person answers, but you I did not hear it ring. 2. Sometimes ringtone is being generated too early. A call is placed from a phone, when it hits *, the caller already hears it ringing, even before asterisk has sent the call out, via IAX or the the PSTN. I dont even now how how to replicate the problems, it just happens, Any suggestions, thanks Hi, guys I have made progress on 1. i.e. I dont hear any ringtone when I dial a number, I just hear the other person start speaking. This is because the device that I am calling sends a SIP message 183 Session Progress. It used to work before, i.e. before when the device used to send SIP message 180 Ringing. Now they have changed the specification on the device to send SIP 183. Does asterisk recognise SIP 183. What can I do to get ringing tone back? Any help is really appreciated thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Aastra 9133i Configurations - are the file namesto be lower case or upper case or does it matter?
Thanks, I did that with upper and lower case, using 1.3. I have another issue then because it is still not loading, it appears the phone is loading but when I check the configs aren't there. Lists wrote: According to the wiki page http://www.voip-info.org/tiki-index.php?page=Aastra+480i+Configuration it shows lowercase file name and then there is a comment at the bottom that it needs to be capitalized. I have tried it both ways with no luck. Could someone comment on which way the cfg files need to be in the /tftpboot directory? Thanks in advance. You need to look up the MAC address of your 9133i. It's on the bar code on the bottom of the phone. If your MAC address looked like this: 00 01 2d 09 58 c1 You would create a file (in uppercase except the .cfg which is in lowercase) called: 00012D0958C1.cfg in the /tftpboot directory. This will configure that SPECIFIC PHONE because it's tied to the MAC address. The case sensitivity only applies to pre-1.3 firmware versions. 1.3 can handle upper and lowercase. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX
Bartosz Piec wrote: Russ Price wrote: So, are there any IP faxes? Sort of. But I'm talking about hardware IP faxes. There are a number of IP capable FAX machines. It seems most don't obey the standards (T.37 and T.38), though. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Forwarding only at certain times
Hi In my extensions.conf shown below when the external number 123 is dialed it goes to phone ext1. I can forward to another phone using exampleline below but I would only like to forward after 5pm and before 9am. How can this be done? Thanks for your help. exten = 123,1,LookupCIDNameexten = 123,2,Dial(SIP/ext1,40)exten = 123,3,Voicemail(1)exten = 123,4,Hangup--- exten = 123,1,goto(XXX,1) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Forwarding only at certain times
James Steven wrote: Hi In my extensions.conf shown below when the external number 123 is dialed it goes to phone ext1. I can forward to another phone using example line below but I would only like to forward after 5pm and before 9am. How can this be done? http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+GotoIfTime Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No application 'MeetMe' for extension
Hi Evert,Do you have the zaptel/ztdummy modules installed ?KunalOn 12/8/05, Evert Meulie [EMAIL PROTECTED] wrote:Hi all!I have an [EMAIL PROTECTED] 2.1 setup here which is working 99% the way it should. The only thing that does not work is Meetme/Conferences... In the log-file I see:Dec 8 11:51:28 WARNING[3288] pbx.c: No application 'MeetMe' for extension (from-internal, 8125, 6)This is when I dial 8125 from extension 125. 8125 is defined in the meetme(-additional).conf And before you ask: yes, ztdummy is loaded...Who has any suggestions?I'm stumped... :-/Regards,Evert___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: No application 'MeetMe' for extension
Read before you reply... ;-) To be 100% clear on zaptel/ztdummy, here's the output of my lsmod: [EMAIL PROTECTED] ~]# lsmod Module Size Used by md5 8001 1 ipv6 240097 16 autofs422085 0 i2c_dev14273 0 i2c_core 25921 1 i2c_dev sunrpc139173 1 ztdummy 7748 0 wctdm 40640 0 wcfxo 16928 0 wcte11xp 30496 0 wct1xxp20768 0 wct4xxp57792 0 tor2 93472 0 zaptel196612 7 ztdummy,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2 crc_ccitt 6081 1 zaptel microcode 11873 0 dm_mirror 28449 0 dm_mod 58949 1 dm_mirror button 10449 0 battery12869 0 ac 8773 0 uhci_hcd 32729 0 ehci_hcd 31813 0 hw_random 9557 0 snd_azx20801 0 snd_hda_codec 75844 1 snd_azx snd_pcm_oss52345 0 snd_mixer_oss 21825 1 snd_pcm_oss snd_pcm91973 3 snd_azx,snd_hda_codec,snd_pcm_oss snd_timer 27973 1 snd_pcm snd56997 6 snd_azx,snd_hda_codec,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer soundcore 12961 1 snd snd_page_alloc 13641 2 snd_azx,snd_pcm 8139too27329 0 mii 8641 1 8139too ext3 118729 2 jbd59481 1 ext3 ata_piix 13253 3 libata 47901 1 ata_piix sd_mod 20545 4 scsi_mod 116429 2 libata,sd_mod Kunal Parikh wrote: Hi Evert, Do you have the zaptel/ztdummy modules installed ? Kunal On 12/8/05, *Evert Meulie* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all! I have an [EMAIL PROTECTED] 2.1 setup here which is working 99% the way it should. The only thing that does not work is Meetme/Conferences... In the log-file I see: Dec 8 11:51:28 WARNING[3288] pbx.c: No application 'MeetMe' for extension (from-internal, 8125, 6) This is when I dial 8125 from extension 125. 8125 is defined in the meetme(-additional).conf And before you ask: yes, ztdummy is loaded... Who has any suggestions? I'm stumped... :-/ Regards, Evert ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multiple line registrations on attendant console
Hi Dionisis, please search the Wiki/ Google for hint in connection with asterisk and you will find. Philipp I've noticed that Polycom and Snom each offer attendant console expansions. As far as I understand, the point in using all thebuttons they provide is to program them to register as extensions in order to be able to monitor the status of each extension at any given time and, also, to be able to pickup any extension that has been ringing for a while. Is this feature supported by asterisk? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call simulators
I'm currently starting development of an add-on to a program designed to be used in a call-centre type environment that will interface very closely with Asterisk - quite possibly to the point that the add-on itself will be a softphone as well. In order to test this application properly, I find myself needing to generate a constant volume of calls to a queue. I can do this by dialling from the two test extensions I have set up on my system, but it would seem a better way of doing this would be to have an external application randomly generate calls at a certain volume. My budget is not big - this is a project for a non-profit volunteer organisation I do a lot of work with so I would obviously prefer something open source. The ability to randomly generate caller ID and intermittently suppress caller ID would be a *very* useful addition. Does anyone know of any software that would fit this bill? If such software doesn't exist, or is beyond my capacity to afford, what other options might I have? My test rig is my home PABX - a very small setup running with three ATAs and two VoIP trunks. It would seem that simulating a trunk would be the best way of doing this, but again, I don't know what is available. Any help would be gratefully received. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Call Recording and SIP canreinvite
Is there a way to optionally keep asterisk in the media path in order to record calls using the Monitor command? For example, if I have a SIP peer that is defined with canreinvite=yes, this means that if possible, Asterisk will not be in the media path. However, what happens if the user presses something like *1 (defined in features.conf) to record the call? Will the call be forced to go through Asterisk automatically? Thanks, Waldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call simulators
Use asterisk itself to build a box which generates the calls. Maybe what some people misses (call simulators are quite a recurrent query on the list) is that you can move a text file with the equivalent of a manager API action Originate in the spool/asterisk/outgoing/ directory and the call will be placed, so it's quite simple to do some intensive test. http://www.asteriskguru.com/tutorials/astertest.html seems nice, never used and I read somewhere it wont compile out of the box with 1.2, but you have the source ... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Call Recording and SIP canreinvite
Is there a way to optionally keep asterisk in the media path in order to record calls using the Monitor command? For example, if I have a SIP peer that is defined with canreinvite=yes, this means that if possible, Asterisk will not be in the media path. However, what happens if the user presses something like *1 (defined in features.conf) to record the call? Will the call be forced to go through Asterisk automatically? Thanks, Waldo I could be wrong but I am pretty sure that once the asterisk is out of the media path then features like *1 will not work since asterisk is not able to listen for it. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Call Recording and SIP canreinvite
If that's the case, is it possible to override the canreinvite attribute of a SIP peer in extensions.conf before a call is made or answered by that peer? - Waldo On Dec 8, 2005, at 9:15 AM, Steve Totaro wrote: Is there a way to optionally keep asterisk in the media path in order to record calls using the Monitor command? For example, if I have a SIP peer that is defined with canreinvite=yes, this means that if possible, Asterisk will not be in the media path. However, what happens if the user presses something like *1 (defined in features.conf) to record the call? Will the call be forced to go through Asterisk automatically? Thanks, Waldo I could be wrong but I am pretty sure that once the asterisk is out of the media path then features like *1 will not work since asterisk is not able to listen for it. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Aastra 9133i Configurations - are the file namesto be lower case or upper case or does it matter?
On Thu, 2005-12-08 at 07:00 -0500, [EMAIL PROTECTED] wrote: Thanks, I did that with upper and lower case, using 1.3. I have another issue then because it is still not loading, it appears the phone is loading but when I check the configs aren't there. How are you checking, with the web interface? If so that's exactly how my 9112i shows up, it is confusing but it has got the settings from my xxx.cfg because everything works, the important thing is does it register, make and receive calls? I did go through a long dialog with Aastra support UK when 1.3 came out. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dynamic IAX2 hosting in the UK
Hi all, just got an iaxy box for a customer and its great, but! I really dont want to host and bill this customer myself and i cannot find a voip to pstn breakout that will let him have a dynamic IP. Gradwell require a static ip Voiptalk wont support it Any Ideas where else to try? Thanks in advance Bails ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dynamic IAX2 hosting in the UK
Hi Bails, We'll help. Drop me a mail off-list. Simon http://www.esms.comOn 12/8/05, bails [EMAIL PROTECTED] wrote: Hi all, just got an iaxy box for a customer and its great, but!I really dont want to host and bill this customer myself and i cannot find a voip to pstn breakout that will let him have a dynamic IP.Gradwell require a static ipVoiptalk wont support itAny Ideas where else to try?Thanks in advanceBails___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Echo cancellation
I am having problems with echo, first let me explain my setup: I have a Gateway box, which basically is an Asterisk with a PRI card. It's only job is to interface with 2 incoming ISDN PRI connections. Then there are two other asterisk boxes to which my users are registered. Dialing from a phone it hits the first asterisk which forwards it to the gateway box and then on to the PSTN. What are the general causes of echo? When calling from my SIP phone I hear no echo but the other end, the PSTN end, hears a lot of echo. What could cause this? Kristian. You need to google echo on the wiki. There are so many causes for echo and possible fixes that work on some installations but not others. Some keywords to search for are echo pri and echo avoidance From reading the list, there is no echo introduced by a PRI and the echo is created by the far side. Nonetheless, it is still your problem. I would first try have a phone register directly to the box with the PRI card and see if there is any difference. I would then check your Zapata.conf settings and adjust gains and also try different echo can settings. Make sure to change one thing at a time and restart asterisk and test. Write down your results so you can get an idea for what is working. Finally, if that is still not helping, you can change the echo can method. Trial and error. Digium sells a hardware echo can upgrade for their cards as well but I think it may only be an option for the quad port cards. Finally, if that still is not working, then you may want to see what 3rd party devices others have used. I have seen success stories posted but am not sure what was used. I think the reason for the 3rd party devices working when * software echo can cannot is the size of the tail. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Recording Volume on Zap Channel
Hi All I have a call center working on 8 FXO Channels, everything working fine except one little problem, I am using asterisk queues with monitor-format = wav49 and monitot-join = yes asterisk is recording all conversations but the problem is that the volume of Zap Channel is too low in most of the calls i am unable to understand what other person was saying (ZAP Channel) although Agent's (SIP Channel) vocie use to get recorded pretty good. Any suggession will be higly appreciated Thanks in Advance Did you try adjusting the gain in Zapata.conf? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE:how to listen voicemail messages
hi all, I have made two voice mail boxes for 2 users on asterisk server (/var/spool/asterisk/voicemail/testmail/inside this 2 boxes for 2 users). i have made following settings in voicemail.conf : [testmail] vipul=,vipul patel, [EMAIL PROTECTED] tejas=,tejas shah,[EMAIL PROTECTED] i have made appprpriate settings in SIP.CONF and EXTENSIONS.CONF. now when any of the user is unavailable voicemail is getting active. It is also allowing to record messages on voicemail box. now my problem is.suppose for one user say TEJAS another user has send voicemail. then how that user TEJAS can listen voicemail message. Is there any command to run on asterisk server. how can i access my voicemail? thanks tejas Create an extension that call voicemailmain. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Odd DTMF issue over PRI
Note: I upgraded Zaptel to the 1.2 stable and changed digits.h line to #define DEFAULT_DTMF_LENGTH 250 * 8. I was told that there is still a problem. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Steven [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] This is an outbound issue that affects SIP and Zap (T1 from another PBX) channels going out our PRI to Telco. I have two ATT conference number that will take the conference access codes. (in theory) (214) 622 4991 (866) 340 2763 If we dial the toll free one, the menus time out because they are not recieving any DTMF. If I wait and connect to the conference receptionist/tech(?) they can do a three way call back in and my DTMF works. (they then tell me there is no problem) If I call the 214 number it works without issue. The odd thing here is that I receive DTMF back from them when it first answers the line. ref: Dec 6 10:28:21 VERBOSE[1448]: -- Called g0/12146224991 Dec 6 10:28:21 DEBUG[1448]: Ooh, format changed from unknown to ulaw Dec 6 10:28:24 DEBUG[1448]: DTMF digit: * on Zap/2-1 Dec 6 10:28:24 DEBUG[1448]: DTMF digit: 8 on Zap/2-1 Dec 6 10:28:24 DEBUG[1448]: Enabled echo cancellation on channel 2 Is this something that they are sending to test/set some DTMF setting on my side, or might I just be hearing them call forward to some other number? The thing that really confuses me is the 866 number. If there is something wrong with my setup, then why does my DTMF work if they 3 way back in. I am still on the same call and do not think any settings on my side would change because of what they do on the other side. But I still think the Issue IS on my side, because if the main toll free ATT Conference number has this problem, I think they would know and would have addressed it. zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,esf,b8zs em=25-48 loadzone = us defaultzone = us zapata.conf: context=from-pstn switchtype=national priindication = outofband signalling=pri_cpe rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=no threewaycalling=no transfer=no cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=no rxgain=0.0 txgain=0.0 faxdetect=no group=0 callgroup=1 pickupgroup=1 immediate=no accountcode=I musiconhold=default channel = 1-23 -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A company that sells Toll Free Number in USA
Alvaro Parres wrote: Hi any one can recommend me a company in the USA that can sell me a Toll Free Number and send me the call via IP. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have had good luck with nufone.. http://nufone.net teliax.com is another good place to look @ -- . -BEGIN GEEK CODE BLOCK- Version: 3.1 GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w-- PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y --END GEEK CODE BLOCK-- . signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hardware combination and type of asterisk configuration
Hi all, I'd like to set up a box with asterisk and the following cards in it : - one E1 card (from digium) - one Junghanns OctoBRI My question is : 1) Is it possible such a configuration ? 2) Because of the Junghanns card, I will have to use the bristuff package, but I'd like to know if this package will also work for the digium card ? Thanks Best regards David Masure ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip behind the NAT
Forward UDP Ports 1-2 to your asterisk box. On 12/8/05, Jeffery Chen [EMAIL PROTECTED] wrote: can u paste your sip.conf general section,,? there have another possible cause... the both side use different codecm and asterisk can not translaste it ... -- Jeffery On 12/8/05, chawki hammoud [EMAIL PROTECTED] wrote: Hi:i added these two lines to my general context ,butnothing happened the same result the sound came in one way for 3 seconds and stopped but it didnt hangup.--- Jeffery Chen [EMAIL PROTECTED] wrote: If your Astersik server behind NAT too, your need modify SIP.conf like this externalIP= x.x.x.x localnet= x.x.x. hope this can help you On 12/8/05, Moises Silva [EMAIL PROTECTED] wrote: what type of NAT do you have? sync? full cone? cone restricted, port restricted? any messages in asterisk verbose console? best regards On 12/7/05, chawki hammoud [EMAIL PROTECTED] wrote: Hi list: i have an asterisk box behind the NAT ,when i try to send calls through Sip to the voip provider server the call is answered but in a one way calling,I hear the voice of the other side just for 4 seconds and then stop but the call do not hangup. my sip.conf is: [voip provider] type=peer host=213.112.50.8 username=XXX secret=XX fromuser=XXX canreinvite=no nat=yes insercure=invite disallow=all allow=gsm __ Yahoo! DSL – Something to write home about. Just $16.99/mo. or less. dsl.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jeffery ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam?Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Nortel Meridian Option81C to TE405P
He said that he is using a crossover but for some reason I think the crossover may be the problem. Try making a new one. Cross pin one with four and two with five. Also try a straight through cable. Your configs look fine on the asterisk side although I am not real cluefull on the Meridian. One question, was the Meridian ever hooked up to the PSTN? Thanks, Steve This might be an obvious question, but should you be using a crossover cable? Information on setting up Nortel to TDM card links can be found at: http://www.pham.org/asterisk/asterisk-meridian-a1.pdf Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Dec 6, 2005, at 2:59 PM, Anish Basu wrote: Hi, I am having problems connecting a Nortel Meridian Option 81C PBX to my Asterisk 1.20 server. We are using the TE405P card with one crossover PRI T1 cable connecting the two systems. The lights on the back of the TE405P are green and zttool shows that the span is okay. Calls cannot be made and 'pri show span 1' shows the d-channel as down. If anyone has any experience with this, suggestions and tips are greatly appreciatd. If we cannot get this resolved within the next few days, we are willing to pay consulting fees for help. The config files are as listed below. Thanks for any help in advance. zaptel.conf --- loadzone = us defaultzone=us span=1,0,0,esf,b8zs bchan=1-23 dchan=24 zapata.conf --- [trunkgroups] [channels] language=en switchtype=5ess context=from-pbx signalling=pri_net group=1 callgroup=1 pickupgroup=1 channel = 1-23 usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 faxdetect=both musiconhold=default Nortel configuration: b-channel,d-channel, and route data block --- REQ prt TYPE adan dch 10 ADAN DCH 10 CTYP MSDL GRP 3 DNUM 2 PORT 0 DES VresaBridge USR PRI DCHL 101 OTBF 32 PARM RS422 DTE DRAT 64KC CLOK EXT IFC ESS5 SIDE USR CNEG 1 RLS ID 1 RCAP ND2 MBGA NO OVLR NO OVLS NO T200 3 T203 10 N200 3 N201 260 K 7 ROUT 1 TYPE RDB CUST 00 ROUT 1 DES VERSA TKTP TIE NPID_TBL_NUM 0 ESN NO CNVT NO SAT NO RCLS EXT VTRK NO DTRK YES BRIP NO DGTP PRI ISDN YES MODE PRA IFC ESS5 SBN NO PNI 1 SRVC NNSF NCNA YES NCRD YES CHTY BCH CTYP UKWN INAC YES ISAR NO CPUB OFF DAPC NO BCOT 0 DSEL VOD PTYP PRI AUTO NO DNIS NO DCDR NO ICOG IAO SRCH LIN TRMB YES STEP ACOD 8901 TCPP NO PII NO TARG 01 CLEN 1 BILN NO OABS INST IDC NO DCNO 0 * NDNO 0 DEXT NO ANTK SIGO STD ICIS YES TIMR ICF 512 OGF 512 EOD 13952 NRD 10112 DDL 70 ODT 4096 RGV 640 GRD 896 SFB 3 NBS 2048 PAGE 002 NBL 4096 IENB 5 TFD 0 VSS 0 VGD 6 DRNG NO CDR NO VRAT NO MUS NO RACD NO FRL 0 0 FRL 1 0 FRL 2 0 FRL 3 0 FRL 4 0 FRL 5 0 FRL 6 0 FRL 7 0 OHQ NO OHQT 00 CBQ NO AUTH NO TDET NO TTBL 0 ATAN NO PLEV 2 ALRM NO ART 0 SGRP 0 AACR NO DES VERSA TN 101 01 TYPE TIE CDEN SD CUST 0 TRK PRI PDCA 1 PCML MU NCOS 0 RTMB 1 73 B-CHANNEL SIGNALING TGAR 1 AST NO IAPG 0 CLS UNR DTN WTA LPR APN THFD HKD P10 VNL TKID DATE 5 DEC 2005 Anish Basu Field Systems Engineer Softel, Inc. Phone: (732) 705-9202 Cell: (732) 312-6634 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] E1/T1 configurations
Hi, If you were to lead someone (with a UI) through the process of configuring a a Digium T1/E1 card with asterisk and a T1/E1 trunk from a provider, would the following questions cover most scenarios? i.e. given the following questions and assumptions, would the configurations below work for most people? - Is it a E1 or T1 line (i.e. are you in Europe or America)? - If it's a T1 line, is it PRI ISDN or EM? - If it's a T1 does it use Extended Super Frame (ESF/D5) framing or is it Super Frame (SF/D4) framing? - If it's an E1, does it use High-density Bipolar-3 (HDB3) or Automatic Mark Inversion (AMI) coding? And given the following assumptions: - If it's an E1 voice line, it's going to be PRI which implies CCS framing - ESF framing on a T1 generally implies B8ZS coding and SF implies AMI coding - If it's a T1 PRI, it's most likely a National 2 switch; if it's an E1 PRI, it's EuroISDN - 0db LBO in most cases - Most E1 lines use a crc - Wink start is usually used with EM signalling You'd then have have a set of configurations like: - T1, PRI, ESF zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 zapata.conf: switchtype=national signalling=pri_cpe context=incoming group=1 channel=1-23 - T1, PRI, SF zaptel.conf: span=1,1,0,d4,ami bchan=1-23 dchan=24 zapata.conf: switchtype=national signalling=pri_cpe context=incoming group=1 channel=1-23 - E1, PRI, HDB3 zaptel.conf: span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 zapata.conf: switchtype=euroisdn signalling=pri_cpe context=incoming group=1 channel=1-23 - E1, PRI, AMI zaptel.conf: span=1,1,0,ccs,ami,crc4 bchan=1-15 dchan=16 bchan=17-31 zapata.conf: switchtype=euroisdn signalling=pri_cpe context=incoming group=1 channel=1-23 - T1, EM, ESF zaptel.conf: span=1,1,0,esf,b8zs em=1-24 zapata.conf: signalling=em_w context=incoming group=1 channel=1-24 - T1, EM, SF zaptel.conf: span=1,1,0,d4,ami em=1-24 zapata.conf: signalling=em_w context=incoming group=1 channel=1-24 Thanks, Mark. Not all E1 providers have crc4 turned on. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nortel Meridian Option81C to TE405P
If the digium card is good then he has the proper cable config, although his send may not be getting to the nortel. This is layer one which must work before layer two, ie d channel. What does the nortel say regarding the T1? If this is good then your issue is with configuration not cableing. On Dec 8, 2005, at 9:02 AM, Steve Totaro wrote: He said that he is using a crossover but for some reason I think the crossover may be the problem. Try making a new one. Cross pin one with four and two with five. Also try a straight through cable. Your configs look fine on the asterisk side although I am not real cluefull on the Meridian. One question, was the Meridian ever hooked up to the PSTN? Thanks, Steve This might be an obvious question, but should you be using a crossover cable? Information on setting up Nortel to TDM card links can be found at: http://www.pham.org/asterisk/asterisk-meridian-a1.pdf Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Dec 6, 2005, at 2:59 PM, Anish Basu wrote: Hi, I am having problems connecting a Nortel Meridian Option 81C PBX to my Asterisk 1.20 server. We are using the TE405P card with one crossover PRI T1 cable connecting the two systems. The lights on the back of the TE405P are green and zttool shows that the span is okay. Calls cannot be made and 'pri show span 1' shows the d-channel as down. If anyone has any experience with this, suggestions and tips are greatly appreciatd. If we cannot get this resolved within the next few days, we are willing to pay consulting fees for help. The config files are as listed below. Thanks for any help in advance. zaptel.conf --- loadzone = us defaultzone=us span=1,0,0,esf,b8zs bchan=1-23 dchan=24 zapata.conf --- [trunkgroups] [channels] language=en switchtype=5ess context=from-pbx signalling=pri_net group=1 callgroup=1 pickupgroup=1 channel = 1-23 usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 faxdetect=both musiconhold=default Nortel configuration: b-channel,d-channel, and route data block --- REQ prt TYPE adan dch 10 ADAN DCH 10 CTYP MSDL GRP 3 DNUM 2 PORT 0 DES VresaBridge USR PRI DCHL 101 OTBF 32 PARM RS422 DTE DRAT 64KC CLOK EXT IFC ESS5 SIDE USR CNEG 1 RLS ID 1 RCAP ND2 MBGA NO OVLR NO OVLS NO T200 3 T203 10 N200 3 N201 260 K7 ROUT 1 TYPE RDB CUST 00 ROUT 1 DES VERSA TKTP TIE NPID_TBL_NUM 0 ESN NO CNVT NO SAT NO RCLS EXT VTRK NO DTRK YES BRIP NO DGTP PRI ISDN YES MODE PRA IFC ESS5 SBN NO PNI 1 SRVC NNSF NCNA YES NCRD YES CHTY BCH CTYP UKWN INAC YES ISAR NO CPUB OFF DAPC NO BCOT 0 DSEL VOD PTYP PRI AUTO NO DNIS NO DCDR NO ICOG IAO SRCH LIN TRMB YES STEP ACOD 8901 TCPP NO PII NO TARG 01 CLEN 1 BILN NO OABS INST IDC NO DCNO 0 * NDNO 0 DEXT NO ANTK SIGO STD ICIS YES TIMR ICF 512 OGF 512 EOD 13952 NRD 10112 DDL 70 ODT 4096 RGV 640 GRD 896 SFB 3 NBS 2048 PAGE 002 NBL 4096 IENB 5 TFD 0 VSS 0 VGD 6 DRNG NO CDR NO VRAT NO MUS NO RACD NO FRL 0 0 FRL 1 0 FRL 2 0 FRL 3 0 FRL 4 0 FRL 5 0 FRL 6 0 FRL 7 0 OHQ NO OHQT 00 CBQ NO AUTH NO TDET NO TTBL 0 ATAN NO PLEV 2 ALRM NO ART 0 SGRP 0 AACR NO DES VERSA TN 101 01 TYPE TIE CDEN SD CUST 0 TRK PRI PDCA 1 PCML MU NCOS 0 RTMB 1 73 B-CHANNEL SIGNALING TGAR 1 AST NO IAPG 0 CLS UNR DTN WTA LPR APN THFD HKD P10 VNL TKID DATE 5 DEC 2005 Anish Basu Field Systems Engineer Softel, Inc. Phone: (732) 705-9202 Cell: (732) 312-6634 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is a BUG ? Hints and incominglimit
i'm using 1.2.
get the right patch from http://bugs.digium.com/view.php?id=5281
patch fie is: Patch-5281-v2.txt
On 12/6/05, Alvaro Parres [EMAIL PROTECTED] wrote:
which version of Asterisk do you have ?, Becouse when i change the function
to your code, every time that one phone with call-limit the Asterisk crash.
I have 1.2.0
On 12/3/05, Paradise Dove [EMAIL PROTECTED] wrote:
hi,
This is the new update_call_counter() which works fine for me:
/*! \brief update_call_counter: Handle call_limit for SIP users
* Note: This is going to be replaced by app_groupcount
* Thought: For realtime, we should propably update storage with inuse
counter... */
static int update_call_counter(struct sip_pvt *fup, int event)
{
char name[256];
int *inuse, *call_limit;
int outgoing = ast_test_flag(fup, SIP_OUTGOING);
struct sip_user *u = NULL;
struct sip_peer *p = NULL;
if (option_debug 2)
ast_log(LOG_DEBUG, Updating call counter for %s call\n,
outgoing ? outgoing : incoming);
/* Test if we need to check call limits, in order to avoid
realtime lookups if we do not need it */
if (!ast_test_flag(fup, SIP_CALL_LIMIT))
return 0;
ast_copy_string(name, fup-username, sizeof(name));
/* Check the list of users */
// paradise dove
p = find_peer(name, NULL, 1);
if (p) {
inuse = p-inUse;
call_limit = p-call_limit;
} else if (!u) {
/* Try to find user */
u = find_user(name, 1);
if (u) {
inuse = u-inUse;
call_limit = u-call_limit;
} else {
if (option_debug 1)
ast_log(LOG_DEBUG, %s is not a local user, no call
limit\n, name);
return 0;
}
}
switch(event) {
/* incoming and outgoing affects the inUse counter */
case DEC_CALL_LIMIT:
if ( *inuse 0 ) {
(*inuse)--;
} else {
*inuse = 0;
}
if (option_debug 1 || sipdebug) {
ast_log(LOG_DEBUG, Call %s %s '%s' removed from call
limit %d\n, outgoing ? to : from, u ? user:peer
}
break;
case INC_CALL_LIMIT:
if (*call_limit 0 ) {
if (*inuse = *call_limit) {
ast_log(LOG_ERROR, Call %s %s '%s' rejected due
to usage limit of %d\n, outgoing ? to : from, u ? u
// paradise dove
if (p)
ASTOBJ_UNREF(p,sip_destroy_peer);
else if (u)
ASTOBJ_UNREF(u,sip_destroy_user);
return -1;
}
}
(*inuse)++;
if (option_debug 1 || sipdebug) {
ast_log(LOG_DEBUG, Call %s %s '%s' is %d out of
%d\n, outgoing ? to : from, u ? user:peer, name, *in
}
break;
default:
ast_log(LOG_ERROR, update_call_counter(%s, %d) called
with no event!\n, name, event);
}
// paradise dove
if (p)
ASTOBJ_UNREF(p,sip_destroy_peer);
else if (u)
ASTOBJ_UNREF(u,sip_destroy_user);
return 0;
}
Paradise Dove
On 12/2/05, Alvaro Parres [EMAIL PROTECTED] wrote:
Could you send it patch please.
On 11/30/05, Paradise Dove [EMAIL PROTECTED] wrote:
btw, i've patched this part of code and now its working fine for me.
i'm going to upload it.
Paradise Dove
On 11/30/05, Kevin Hanson [EMAIL PROTECTED] wrote:
Paradise Dove wrote:
Yes with version 1.2. I have tried already with call-limit and the
same.
i agree with you, it seems to be a bug which i've submited before
(bug
#5281) but it's now closed by bug marshals!
It's not closed. It's suspended waiting input from you:
Closing until the appropriate debug/trace output can be provided.
On 10/30 you said you were still trying to get the debug output.
Cheers,
Kevin
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[Asterisk-Users] Octo Bri card together te405p and bristuff
Hi, is it possible to run an octoBri card together with a TE405P card in one system with bristuff? If yes, how should the zaptel.conf look like? Thanks and regards, BK ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nortel Meridian Option81C to TE405P
You can't use a ethernet crossover cable, make sure you are using a T1 crossover cable. (you will definately need to use a T1 crossover cable). I'm running a Nortel Option 11 and Asterisk connected in this manner. On 12/8/05, Steve Totaro [EMAIL PROTECTED] wrote: He said that he is using a crossover but for some reason I think thecrossover may be the problem.Try making a new one.Cross pin one with four and two with five.Also try a straight through cable.Yourconfigs look fine on the asterisk side although I am not real cluefullon the Meridian.One question, was the Meridian ever hooked up to the PSTN? Thanks,Steve This might be an obvious question, but should you be using a crossover cable? Information on setting up Nortel to TDM card links can be found at: http://www.pham.org/asterisk/asterisk-meridian-a1.pdf Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Dec 6, 2005, at 2:59 PM, Anish Basu wrote: Hi, I am having problems connecting a Nortel Meridian Option 81C PBX tomy Asterisk 1.20 server. We are using the TE405P card with onecrossover PRI T1 cable connecting the two systems. The lights on the back of the TE405P are green and zttool shows that the span is okay. Calls cannot be made and 'pri show span 1' shows the d-channel as down. If anyone has any experience with this, suggestions and tips are greatly appreciatd. If wecannot get this resolved within the next few days, we are willing to pay consulting fees for help. The config files are as listed below. Thanks forany help in advance.zaptel.conf --- loadzone = us defaultzone=us span=1,0,0,esf,b8zs bchan=1-23 dchan=24 zapata.conf --- [trunkgroups] [channels] language=en switchtype=5ess context=from-pbx signalling=pri_net group=1 callgroup=1 pickupgroup=1 channel = 1-23 usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 faxdetect=both musiconhold=default Nortel configuration: b-channel,d-channel, and route data block --- REQ prt TYPE adan dch 10 ADAN DCH 10 CTYP MSDL GRP 3 DNUM 2 PORT 0 DES VresaBridge USR PRI DCHL 101 OTBF 32 PARM RS422 DTE DRAT 64KC CLOK EXT IFC ESS5 SIDE USR CNEG 1 RLS ID 1 RCAP ND2 MBGA NO OVLR NO OVLS NO T200 3 T203 10 N200 3 N201 260 K 7ROUT 1 TYPE RDB CUST 00 ROUT 1 DES VERSA TKTP TIE NPID_TBL_NUM 0 ESN NO CNVT NO SAT NO RCLS EXT VTRK NO DTRK YES BRIP NO DGTP PRI ISDN YES MODE PRA IFC ESS5 SBN NO PNI 1 SRVC NNSF NCNA YES NCRD YES CHTY BCH CTYP UKWN INAC YES ISAR NO CPUB OFF DAPC NO BCOT 0 DSEL VOD PTYP PRI AUTO NO DNIS NO DCDR NO ICOG IAO SRCH LIN TRMB YES STEP ACOD 8901 TCPP NO PII NO TARG 01 CLEN 1 BILN NO OABS INST IDC NO DCNO 0 * NDNO 0 DEXT NO ANTK SIGO STD ICIS YES TIMR ICF 512 OGF 512 EOD 13952 NRD 10112 DDL 70 ODT 4096 RGV 640 GRD 896 SFB 3 NBS 2048PAGE 002 NBL 4096 IENB 5 TFD 0 VSS 0 VGD 6 DRNG NO CDR NO VRAT NO MUS NO RACD NO FRL 0 0 FRL 1 0 FRL 2 0 FRL 3 0 FRL 4 0 FRL 5 0 FRL 6 0 FRL 7 0 OHQ NO OHQT 00 CBQ NO AUTH NO TDET NO TTBL 0 ATAN NO PLEV 2 ALRM NO ART 0 SGRP 0 AACR NO DES VERSA TN 101 01 TYPE TIE CDEN SD CUST 0 TRK PRI PDCA 1 PCML MU NCOS 0 RTMB 1 73 B-CHANNEL SIGNALING TGAR 1 AST NO IAPG 0 CLS UNR DTN WTA LPR APN THFD HKD P10 VNL TKID DATE 5 DEC 2005 Anish Basu Field Systems Engineer Softel, Inc. Phone: (732) 705-9202 Cell: (732) 312-6634 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Nortel Meridian Option81C to TE405P
Cabling is always the first thing to check in these types of issues. Sure, he may need a T1 crossover but maybe the cable was made incorrectly or the connections are not crimped well. Just because someone says they have a crossover cable does not mean that it is OK. Again, check the cable (physical layer one) Thanks, Steve If the digium card is good then he has the proper cable config, although his send may not be getting to the nortel. This is layer one which must work before layer two, ie d channel. What does the nortel say regarding the T1? If this is good then your issue is with configuration not cableing. On Dec 8, 2005, at 9:02 AM, Steve Totaro wrote: He said that he is using a crossover but for some reason I think the crossover may be the problem. Try making a new one. Cross pin one with four and two with five. Also try a straight through cable. Your configs look fine on the asterisk side although I am not real cluefull on the Meridian. One question, was the Meridian ever hooked up to the PSTN? Thanks, Steve This might be an obvious question, but should you be using a crossover cable? Information on setting up Nortel to TDM card links can be found at: http://www.pham.org/asterisk/asterisk-meridian-a1.pdf Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Dec 6, 2005, at 2:59 PM, Anish Basu wrote: Hi, I am having problems connecting a Nortel Meridian Option 81C PBX to my Asterisk 1.20 server. We are using the TE405P card with one crossover PRI T1 cable connecting the two systems. The lights on the back of the TE405P are green and zttool shows that the span is okay. Calls cannot be made and 'pri show span 1' shows the d-channel as down. If anyone has any experience with this, suggestions and tips are greatly appreciatd. If we cannot get this resolved within the next few days, we are willing to pay consulting fees for help. The config files are as listed below. Thanks for any help in advance. zaptel.conf --- loadzone = us defaultzone=us span=1,0,0,esf,b8zs bchan=1-23 dchan=24 zapata.conf --- [trunkgroups] [channels] language=en switchtype=5ess context=from-pbx signalling=pri_net group=1 callgroup=1 pickupgroup=1 channel = 1-23 usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 faxdetect=both musiconhold=default Nortel configuration: b-channel,d-channel, and route data block --- REQ prt TYPE adan dch 10 ADAN DCH 10 CTYP MSDL GRP 3 DNUM 2 PORT 0 DES VresaBridge USR PRI DCHL 101 OTBF 32 PARM RS422 DTE DRAT 64KC CLOK EXT IFC ESS5 SIDE USR CNEG 1 RLS ID 1 RCAP ND2 MBGA NO OVLR NO OVLS NO T200 3 T203 10 N200 3 N201 260 K7 ROUT 1 TYPE RDB CUST 00 ROUT 1 DES VERSA TKTP TIE NPID_TBL_NUM 0 ESN NO CNVT NO SAT NO RCLS EXT VTRK NO DTRK YES BRIP NO DGTP PRI ISDN YES MODE PRA IFC ESS5 SBN NO PNI 1 SRVC NNSF NCNA YES NCRD YES CHTY BCH CTYP UKWN INAC YES ISAR NO CPUB OFF DAPC NO BCOT 0 DSEL VOD PTYP PRI AUTO NO DNIS NO DCDR NO ICOG IAO SRCH LIN TRMB YES STEP ACOD 8901 TCPP NO PII NO TARG 01 CLEN 1 BILN NO OABS INST IDC NO DCNO 0 * NDNO 0 DEXT NO ANTK SIGO STD ICIS YES TIMR ICF 512 OGF 512 EOD 13952 NRD 10112 DDL 70 ODT 4096 RGV 640 GRD 896 SFB 3 NBS 2048 PAGE 002 NBL 4096 IENB 5 TFD 0 VSS 0 VGD 6 DRNG NO CDR NO VRAT NO MUS NO RACD NO FRL 0 0 FRL 1 0 FRL 2 0 FRL 3 0 FRL 4 0 FRL 5 0 FRL 6 0 FRL 7 0 OHQ NO OHQT 00 CBQ NO AUTH NO TDET NO TTBL 0 ATAN NO PLEV 2 ALRM NO ART 0 SGRP 0 AACR NO DES VERSA TN 101 01 TYPE TIE CDEN SD CUST 0 TRK PRI PDCA 1 PCML MU NCOS 0 RTMB 1 73 B-CHANNEL SIGNALING TGAR 1 AST NO IAPG 0 CLS UNR DTN WTA LPR APN THFD HKD P10 VNL TKID DATE 5 DEC 2005 Anish Basu Field Systems Engineer Softel, Inc. Phone: (732) 705-9202 Cell: (732) 312-6634 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom SIP part numbers
Greetings All, I intend to buy a Polycom IP 500CS with part number 2201.11500.001 but I'm not sure if it will work with Asterisk. Is this a SIP capable phone? Moreover, what does CS stand for? Done some research at http://www.voip-info.org/wiki/view/Polycom+Phones but the original info on part numbers and supported protocols seems to have been erased. Note that I currently have a Polycom IP300 SIP phone with part number 2201.11300.001 and it works perfectly with Asterisk. Please assist if you have a phone similar to the one mentioned above. -- Thanks regards, Alphonse Ogulla Nairobi, Kenya ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Call Recording and SIP canreinvite
Well, then set canreinvite=no If that's the case, is it possible to override the canreinvite attribute of a SIP peer in extensions.conf before a call is made or answered by that peer? - Waldo On Dec 8, 2005, at 9:15 AM, Steve Totaro wrote: Is there a way to optionally keep asterisk in the media path in order to record calls using the Monitor command? For example, if I have a SIP peer that is defined with canreinvite=yes, this means that if possible, Asterisk will not be in the media path. However, what happens if the user presses something like *1 (defined in features.conf) to record the call? Will the call be forced to go through Asterisk automatically? Thanks, Waldo I could be wrong but I am pretty sure that once the asterisk is out of the media path then features like *1 will not work since asterisk is not able to listen for it. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A2billing Areskicc Incoming DID setting
Anyone know how to do it? Sam ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Call simulators
Hello Rob, Our OrderlyQ system is designed to pass (real) calls to call centre agents and queues at a constant rate (or at least can easily be configured to do this). I can think of several ways the system could be 'rigged' to produce the calls automatically too... We've also built our own call centre simulators as part of the development effort for OrderlyQ. Let me know if we can help, Matt King, M.A. Oxon. http://www.orderlyq.com - the world's most advanced queue system. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E1/T1 configurations
Steve Totaro wrote: Hi, If you were to lead someone (with a UI) through the process of configuring a a Digium T1/E1 card with asterisk and a T1/E1 trunk from a provider, would the following questions cover most scenarios? i.e. given the following questions and assumptions, would the configurations below work for most people? - Is it a E1 or T1 line (i.e. are you in Europe or America)? - If it's a T1 line, is it PRI ISDN or EM? - If it's a T1 does it use Extended Super Frame (ESF/D5) framing or is it Super Frame (SF/D4) framing? - If it's an E1, does it use High-density Bipolar-3 (HDB3) or Automatic Mark Inversion (AMI) coding? And given the following assumptions: - If it's an E1 voice line, it's going to be PRI which implies CCS framing - ESF framing on a T1 generally implies B8ZS coding and SF implies AMI coding - If it's a T1 PRI, it's most likely a National 2 switch; if it's an E1 PRI, it's EuroISDN - 0db LBO in most cases - Most E1 lines use a crc - Wink start is usually used with EM signalling You'd then have have a set of configurations like: - T1, PRI, ESF zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 zapata.conf: switchtype=national signalling=pri_cpe context=incoming group=1 channel=1-23 - T1, PRI, SF zaptel.conf: span=1,1,0,d4,ami bchan=1-23 dchan=24 zapata.conf: switchtype=national signalling=pri_cpe context=incoming group=1 channel=1-23 - E1, PRI, HDB3 zaptel.conf: span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 zapata.conf: switchtype=euroisdn signalling=pri_cpe context=incoming group=1 channel=1-23 - E1, PRI, AMI zaptel.conf: span=1,1,0,ccs,ami,crc4 bchan=1-15 dchan=16 bchan=17-31 zapata.conf: switchtype=euroisdn signalling=pri_cpe context=incoming group=1 channel=1-23 - T1, EM, ESF zaptel.conf: span=1,1,0,esf,b8zs em=1-24 zapata.conf: signalling=em_w context=incoming group=1 channel=1-24 - T1, EM, SF zaptel.conf: span=1,1,0,d4,ami em=1-24 zapata.conf: signalling=em_w context=incoming group=1 channel=1-24 Thanks, Mark. Not all E1 providers have crc4 turned on. Except for ISDN lines, E1s hardly ever have CRC4 turned on. As you said, some countries don't even turn it on for ISDN, which is stupid. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Nortel Meridian Option81C to TE405P
Yes, that is why I said cross pins one with four and two with five (a T1 crossover cable configuration) You can't use a ethernet crossover cable, make sure you are using a T1 crossover cable. (you will definately need to use a T1 crossover cable). I'm running a Nortel Option 11 and Asterisk connected in this manner. On 12/8/05, Steve Totaro [EMAIL PROTECTED] wrote: He said that he is using a crossover but for some reason I think the crossover may be the problem. Try making a new one. Cross pin one with four and two with five. Also try a straight through cable. Your configs look fine on the asterisk side although I am not real cluefull on the Meridian. One question, was the Meridian ever hooked up to the PSTN? Thanks, Steve This might be an obvious question, but should you be using a crossover cable? Information on setting up Nortel to TDM card links can be found at: http://www.pham.org/asterisk/asterisk-meridian-a1.pdf Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Dec 6, 2005, at 2:59 PM, Anish Basu wrote: Hi, I am having problems connecting a Nortel Meridian Option 81C PBX to my Asterisk 1.20 server. We are using the TE405P card with one crossover PRI T1 cable connecting the two systems. The lights on the back of the TE405P are green and zttool shows that the span is okay. Calls cannot be made and 'pri show span 1' shows the d-channel as down. If anyone has any experience with this, suggestions and tips are greatly appreciatd. If we cannot get this resolved within the next few days, we are willing to pay consulting fees for help. The config files are as listed below. Thanks for any help in advance. zaptel.conf --- loadzone = us defaultzone=us span=1,0,0,esf,b8zs bchan=1-23 dchan=24 zapata.conf --- [trunkgroups] [channels] language=en switchtype=5ess context=from-pbx signalling=pri_net group=1 callgroup=1 pickupgroup=1 channel = 1-23 usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 faxdetect=both musiconhold=default Nortel configuration: b-channel,d-channel, and route data block --- REQ prt TYPE adan dch 10 ADAN DCH 10 CTYP MSDL GRP 3 DNUM 2 PORT 0 DES VresaBridge USR PRI DCHL 101 OTBF 32 PARM RS422 DTE DRAT 64KC CLOK EXT IFC ESS5 SIDE USR CNEG 1 RLS ID 1 RCAP ND2 MBGA NO OVLR NO OVLS NO T200 3 T203 10 N200 3 N201 260 K 7 ROUT 1 TYPE RDB CUST 00 ROUT 1 DES VERSA TKTP TIE NPID_TBL_NUM 0 ESN NO CNVT NO SAT NO RCLS EXT VTRK NO DTRK YES BRIP NO DGTP PRI ISDN YES MODE PRA IFC ESS5 SBN NO PNI 1 SRVC NNSF NCNA YES NCRD YES CHTY BCH CTYP UKWN INAC YES ISAR NO CPUB OFF DAPC NO BCOT 0 DSEL VOD PTYP PRI AUTO NO DNIS NO DCDR NO ICOG IAO SRCH LIN TRMB YES STEP ACOD 8901 TCPP NO PII NO TARG 01 CLEN 1 BILN NO OABS INST IDC NO DCNO 0 * NDNO 0 DEXT NO ANTK SIGO STD ICIS YES TIMR ICF 512 OGF 512 EOD 13952 NRD 10112 DDL 70 ODT 4096 RGV 640 GRD 896 SFB 3 NBS 2048 PAGE 002 NBL 4096 IENB 5 TFD 0 VSS 0 VGD 6 DRNG NO CDR NO VRAT NO MUS NO RACD
Re: [Asterisk-Users] Asterisk Call Recording and SIP canreinvite
I understand. But because the majority of calls are not to be recorded, I don't have a need to keep Asterisk in the media path all the time. That's why I'm wondering if you could dynamically keep it in the media path or not. - Waldo On Dec 8, 2005, at 10:26 AM, Steve Totaro wrote: Well, then set canreinvite=no If that's the case, is it possible to override the canreinvite attribute of a SIP peer in extensions.conf before a call is made or answered by that peer? - Waldo On Dec 8, 2005, at 9:15 AM, Steve Totaro wrote: Is there a way to optionally keep asterisk in the media path in order to record calls using the Monitor command? For example, if I have a SIP peer that is defined with canreinvite=yes, this means that if possible, Asterisk will not be in the media path. However, what happens if the user presses something like *1 (defined in features.conf) to record the call? Will the call be forced to go through Asterisk automatically? Thanks, Waldo I could be wrong but I am pretty sure that once the asterisk is out of the media path then features like *1 will not work since asterisk is not able to listen for it. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Aastra 9133i Configurations - are the file namesto be lower case or upper case or does it matter?
[EMAIL PROTECTED] wrote: Thanks, I did that with upper and lower case, using 1.3. I have another issue then because it is still not loading, it appears the phone is loading but when I check the configs aren't there. I looked at this last night. You need to have an aastra.cfg file in your directory in addition to the mac.cfg. If the aastra.cfg file is not present, your phone will not download anything. This is a bug that was supposedly fixed in 1.3 but I found that it still is broken. So, add the aastra.cfg. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New GSM 1-8 ports Gateway / Terminal for sale (with SMS Feature and Many more)
The long waited Ultimate GSM Gateway is finally out. This time we have managed to source a new patch of brand NEW GSM Gateway at prices that is only 50% of what the market rate. And with the SMS Function and many more... For purchase please email gsm AT cyper-telecom.net. We accept paypal and bank transfer. Postage is not included. Please notice we have also got the standard Dual Band GSM gateway for £60 per unit. Introduction: Cyber-Telecom Fixed Wireless GSM Gateway Devices integrated GSM and CDMA technologies. Features: Security features including terminal lock, SIM lock, Carrier Lock and District Lock Supports least cost routing based phone # dialled Network Management through SMS: can configure FWT parameters and query FWT parameters Parameter management: parameters can be modified either by phone or via NMS software Billing signal support by providing reverse polarity signals Models: GSM-TRI-SMS-01 RJ11 interface 1 GSM/CDMA interface £99 per unit GSM-TRI-SMS-04 4 RJ11 interfaces 4 GSM interfaces £399 per unit GSM-TRI-SMS-08 8 RJ11 interfaces 8 GSM interfaces £799 per unit Technical Specifications: Working spectrumGSM900/GSM1800/GSM1900MHz CDMA800/CDMA1900MHz Power AC 110-220V/50Hz Temperature -20 ℃ - 40℃ Humidity10%-95% ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Exit Voicemail
Is there a way to have control go back to the dialplan after a call gets tovoicemail? I'm looking to implement findme and campon, but I wantthe options to be hidden, so if someone calling got a voicemail they could key in *1 (or whatever) and it would go back to the dialplan so I can implement finemein the dial plan. The same with campon, if you got a busy voicemail you could key in *2 (or whatever) and it would take them to the piece of the dialplan where it would wait for person to get off the phone. I realize I could do this by having the user key in another option (Hit 1 to leave a voicemail, hit 2 to findme) but would prefer not to, users could record this as part of their voicemail message if they want the public to know about the findme and camping on a busy extension. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New GSM 1-8 ports Gateway / Terminal for sale (with SMS Feature and Many more)
Well up until I saw the 100-220V and 50HZ I was sold.. But if you dont support 60HZ it will never work in North America. Well it could but it would be a pain in the ass.. /b On Thu, 2005-12-08 at 23:41 +0800, Sam Tam wrote: The long waited Ultimate GSM Gateway is finally out. This time we have managed to source a new patch of brand NEW GSM Gateway at prices that is only 50% of what the market rate. And with the SMS Function and many more... For purchase please email gsm AT cyper-telecom.net. We accept paypal and bank transfer. Postage is not included. Please notice we have also got the standard Dual Band GSM gateway for £60 per unit. Introduction: Cyber-Telecom Fixed Wireless GSM Gateway Devices integrated GSM and CDMA technologies. Features: Security features including terminal lock, SIM lock, Carrier Lock and District Lock Supports least cost routing based phone # dialled Network Management through SMS: can configure FWT parameters and query FWT parameters Parameter management: parameters can be modified either by phone or via NMS software Billing signal support by providing reverse polarity signals Models: GSM-TRI-SMS-01 RJ11 interface 1 GSM/CDMA interface £99 per unit GSM-TRI-SMS-04 4 RJ11 interfaces 4 GSM interfaces £399 per unit GSM-TRI-SMS-08 8 RJ11 interfaces 8 GSM interfaces £799 per unit Technical Specifications: Working spectrum GSM900/GSM1800/GSM1900MHz CDMA800/CDMA1900MHz Power AC 110-220V/50Hz Temperature -20 ℃ - 40℃ Humidity 10%-95% ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _.._ Brian Fertig Data/Telecom Engineer IT Administrator Planet Telecom, Inc Tampa, FL Office o: +1.813.864.3161x107 f: +1.813.881.9762 d: +1.813.864.3164 SIP URI: [EMAIL PROTECTED] This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Call Recording and SIP canreinvite
On 12/8/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: I understand. But because the majority of calls are not to be recorded, I don't have a need to keep Asterisk in the media path all the time. That's why I'm wondering if you could dynamically keep it in the media path or not. Some options of the Dial command force * to stay in the media path, like t (to let user transfer by hitting #). So you could just put one of thos options in your dial string hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Call Recording and SIP canreinvite
There may be a better way but off the top of my head this idea jumped out. It assumes that you know prior to making the call that you need to record it and that you have phones capable of multiple lines. Setup a second line with a different entry in sip.conf with canreinvite=no and use that line to make your calls. Other than that I see reference on the wiki to an H option in dial but have never used it. I think you will still need to know prior to dialing whether you will want to record the call or not so you can dial the exten that uses the H option. If you get this to work, please post your results back to this thread. Re: Re: H option by flobi on Monday 25 of July, 2005 [10:43:46] why not just set canreinvite=yes and on the calls where you don't want reinvite use the H option (if it actually does disable reinvite) or the T or t which also disable reinvite. 7960G Seems to need canreinvite=no as well. by Anonymous on Friday 29 of October, 2004 [22:22:43] Running P0S3-07-2-00. Re: H option by Anonymous on Monday 26 of July, 2004 [10:10:07] (:confused:) Hmm... Now I started to wonder, if it's somehow possible to override the canreinvite=no setting on per call basis. Anyone? H option by Anonymous on Saturday 10 of July, 2004 [04:15:13] Asterisk will not reinvite if the H option is used in the Dial command. http://www.voip-info.org/wiki-Asterisk+sip+canreinvite Thanks, Steve I understand. But because the majority of calls are not to be recorded, I don't have a need to keep Asterisk in the media path all the time. That's why I'm wondering if you could dynamically keep it in the media path or not. - Waldo On Dec 8, 2005, at 10:26 AM, Steve Totaro wrote: Well, then set canreinvite=no If that's the case, is it possible to override the canreinvite attribute of a SIP peer in extensions.conf before a call is made or answered by that peer? - Waldo On Dec 8, 2005, at 9:15 AM, Steve Totaro wrote: Is there a way to optionally keep asterisk in the media path in order to record calls using the Monitor command? For example, if I have a SIP peer that is defined with canreinvite=yes, this means that if possible, Asterisk will not be in the media path. However, what happens if the user presses something like *1 (defined in features.conf) to record the call? Will the call be forced to go through Asterisk automatically? Thanks, Waldo I could be wrong but I am pretty sure that once the asterisk is out of the media path then features like *1 will not work since asterisk is not able to listen for it. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zombie AGI processes in FC2 / 1.2 Beta 1 under l oad
I have an extremely simple AGI script like this: #!/bin/bash ISODATE=`date -iso-8601=seconds` echo SET VARIABLE ISODATE \$ISODATE\ The script's intent is to return the current date and time back to Asterisk in an ISO format as a variable. It works fine. Calling the script from Asterisk returns the variable correctly. Under load ( a dozen or so concurrent calls) the script still works but it tends to zombie. Sometimes it does, sometimes it doesn't. I've seen some background: http://lists.digium.com/pipermail/asterisk-users/2003-August/017310.html http://lists.digium.com/pipermail/asterisk-users/2003-August/017202.html But this is FC2, latest SMP kernel / 1.2 Beta 1 . The zombies build up and appear not to affect call quality but I'd like to eliminate them. Anyone have any tips? tia ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OOH323 towards cisco gateway (2691) call setup fails at q931: Mandatory information element is missing (96)
Hi, I am using ooh323. I cannot setup a call towards a cisco gateway. The cisco rejects the call right away with : Cause value: Mandatory information element is missing (96) This is in the q931 part. Cisco explanation Indicates that the equipment that is sending this code has received a message that is missing an information element that must be present in the message before that message can be processed. Show version gives : Cvs-head-06/21/05-23:51:26 Someone any clue ? H323.conf : ; Objective System's H323 Configuration example for Asterisk ; ooh323c driver configuration ; ; [general] section defines global parameters ; ; This is followed by profiles which can be of three types - user/peer/friend ; Name of the user profile should match with the h323id of the user device. ; For peer/friend profiles, host ip address must be provided as dynamic is ; not supported as of now. ; ; Syntax for specifying a H323 device in extensions.conf is ; For Registered peers/friends profiles: ; H323/name where name is the name of the peer/friend profile. ; ; For unregistered H.323 phones: ; H323/ip[:port] OR if gk is used H323/alias where alias can be any H323 ; alias ; ; For dialing into another asterisk peer at a specific exten ; H323/exten/peer OR H323/[EMAIL PROTECTED] ; ; Domain name resolution is not yet supported. ; ; When a H.323 user calls into asterisk, his H323ID is matched with the profile ; name and context is determined to route the call ; ; The channel driver will register all global aliases and aliases defined in ; peer profiles with the gatekeeper, if one exists. So, that when someone ; outside our pbx (non-user) calls an extension, gatekeeper will route that ; call to our asterisk box, from where it will be routed as per dial plan. [general] ;Define the asetrisk server h323 endpoint ;The port asterisk should listen for incoming H323 connections. ;Default - 1720 port=1720 ;The dotted IP address asterisk should listen on for incoming H323 ;connections ;Default - tries to find out local ip address on it's own bindaddr=0.0.0.0 ;UPDATE this to proper ip address of your asterisk box ;Whether asterisk should use fast-start and tunneling for H323 connections. ;Default - yes faststart=yes h245tunneling=yes ;H323-ID to be used for asterisk server ;Default - Asterisk PBX h323id=TK_BRU_AST1 e164=100 ;CallerID to use for calls ;Default - Same as h323id callerid=TK_BRU_AST1 ;Whether this asterisk server will use gatekeeper. ;Default - DISABLE ;gatekeeper = DISCOVER ;gatekeeper = a.b.c.d gatekeeper = DISABLE ;Location for H323 log file ;Default - /var/log/asterisk/h323_log logfile=/var/log/asterisk/h323_log ;Following values apply to all users/peers/friends defined below, unless ;overridden within their client definition ;Sets default context all clients will be placed in. ;Default - default context=from-sip2 ;Sets rtptimeout for all clients, unless overridden ;Default - 60 seconds ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity ; when we're not on hold ;Type of Service ;Default - none (lowdelay, thoughput, reliability, mincost, none) ;tos=lowdelay ;amaflags = default ;The account code used by default for all clients. ;accountcode=h3230101 ;The codecs to be used for all clients. ;Default - ulaw ; ONLY ulaw, alaw, gsm, g729 and g723 (g723.1) are supported as of now disallow=all ;Note order of disallow/allow is important. allow=g729 allow=alaw allow=ulaw ; dtmf mode to be used by default for all clients. Only rfc2833 supported as ; of now. ;Default - rfc 2833 dtmfmode=rfc2833 ; User/peer/friend definitions: [TK_BRU_GW1] type=friend context=from-sip2 ip=195.xxx.yyy.zzz port=1720 disallow=all allow=g729 incominglimit=3 outgoinglimit=3 rtptimeout=60 dtmfmode=rfc2833 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.13.12/193 - Release Date: 6/12/2005 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Maximum Calls handled
I have a big dilemma. I have a client who is looking for a big installation. I am looking at the digium product and have the following Questions. Difference between Asterisk and Asterisk Business Edition. My Client has 300 personal split between two office and wants to use one asterisk box to support those calls. He is going to have 3 PRIS coming in. Can I use the regular version of Asterisk compared to the Business Edition of Asterisk. How many simultaneous calls can Asterisk support compared to the Business Edition of Asterisk. Please help me out as I dont want to make the wrong recommendations. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New GSM 1-8 ports Gateway / Terminal for sale(with SMS Feature and Many more)
What will work in the US and also have SMS with multiple ports? Well up until I saw the 100-220V and 50HZ I was sold.. But if you dont support 60HZ it will never work in North America. Well it could but it would be a pain in the ass.. /b On Thu, 2005-12-08 at 23:41 +0800, Sam Tam wrote: The long waited Ultimate GSM Gateway is finally out. This time we have managed to source a new patch of brand NEW GSM Gateway at prices that is only 50% of what the market rate. And with the SMS Function and many more... For purchase please email gsm AT cyper-telecom.net. We accept paypal and bank transfer. Postage is not included. Please notice we have also got the standard Dual Band GSM gateway for £60 per unit. Introduction: Cyber-Telecom Fixed Wireless GSM Gateway Devices integrated GSM and CDMA technologies. Features: Security features including terminal lock, SIM lock, Carrier Lock and District Lock Supports least cost routing based phone # dialled Network Management through SMS: can configure FWT parameters and query FWT parameters Parameter management: parameters can be modified either by phone or via NMS software Billing signal support by providing reverse polarity signals Models: GSM-TRI-SMS-01 RJ11 interface 1 GSM/CDMA interface £99 per unit GSM-TRI-SMS-04 4 RJ11 interfaces 4 GSM interfaces £399 per unit GSM-TRI-SMS-08 8 RJ11 interfaces 8 GSM interfaces £799 per unit Technical Specifications: Working spectrumGSM900/GSM1800/GSM1900MHz CDMA800/CDMA1900MHz Power AC 110-220V/50Hz Temperature -20 ℃ - 40℃ Humidity10%-95% ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _.._ Brian Fertig Data/Telecom Engineer IT Administrator Planet Telecom, Inc Tampa, FL Office o: +1.813.864.3161x107 f: +1.813.881.9762 d: +1.813.864.3164 SIP URI: [EMAIL PROTECTED] This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail context
Hello, In the process of upgrading a couple of voicemail servers from CVS (end of august 2005) to 1.2.1 This is a purely voicemail system using mysql configurations. All my mailboxes are in the default context and it worked fine under the CVS version. But with 1.2.1 the voicemailmain fails to authenticate. I debugged and searched around a bit. And found the problem in the Mysql request. Under CVS the request was: Dec 8 10:13:53 DEBUG[32760] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM users WHERE mailbox = '201' AND context = 'default' Under 1.2.1 the request is: Dec 7 15:09:55 DEBUG[3900] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM users WHERE mailbox = '201' AND context = '' According to the documentation, if I don't specify the context, it should take the default context shouldn't it? After googling a bit, I found bug 5899 which seems to be related, but the way I understand it, the old behaviour was if no context pas specified, it selected without context. In the new version it selects the default context. Tried testing with the searchcontexts=yes. It recognises the mailbox, but doesn't seem to match the password. Dec 8 10:41:34 VERBOSE[5567] logger.c: -- Executing VoiceMailMain(SIP/-c37a, 201) in new stack Dec 8 10:41:34 DEBUG[5567] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM users WHERE mailbox = '201' Dec 8 10:41:34 DEBUG[5567] res_config_mysql.c: MySQL RealTime: Everything is fine. Dec 8 10:41:34 VERBOSE[5567] logger.c: -- Playing 'vm-password' (language 'en') Dec 8 10:41:37 VERBOSE[5567] logger.c: -- Incorrect password '123' for user '201' (context = default) Am I missing something? When I don't specify the context, it should take the default context or the default context ? Where do I specify it? Any ideas why the password match isin't working if I revert to the old method ? Example mysql entry: ++---+---+---+++-+-+ --+--+--++-+--+---+ |uniqueid|customer_id|context|mailbox|password|fullname|email |pager|attach|saycid|delete|envelope|serveremail |stamp |options| ++---+---+---+++-+-+ --+--+--++-+--+---+ | 1| 0|default| 201 | 123|Bob |[EMAIL PROTECTED]| |yes |no|no|no |test@test.com|20051207164443|NULL | ++---+---+---+++-+-+ --+--+--++-+--+---+ Extension is simply a Voicemailmain(201) Thanks, Benjamin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Call Recording and SIP canreinvite
This and Time Bandit's comment makes sense. I didn't realize that these options in the Dial string will force Asterisk to stay in the media path even if canreinvite=yes. I'll give it a try. Thanks, Waldo On Dec 8, 2005, at 11:18 AM, Steve Totaro wrote: There may be a better way but off the top of my head this idea jumped out. It assumes that you know prior to making the call that you need to record it and that you have phones capable of multiple lines. Setup a second line with a different entry in sip.conf with canreinvite=no and use that line to make your calls. Other than that I see reference on the wiki to an H option in dial but have never used it. I think you will still need to know prior to dialing whether you will want to record the call or not so you can dial the exten that uses the H option. If you get this to work, please post your results back to this thread. Re: Re: H option by flobi on Monday 25 of July, 2005 [10:43:46] why not just set canreinvite=yes and on the calls where you don't want reinvite use the H option (if it actually does disable reinvite) or the T or t which also disable reinvite. 7960G Seems to need canreinvite=no as well. by Anonymous on Friday 29 of October, 2004 [22:22:43] Running P0S3-07-2-00. Re: H option by Anonymous on Monday 26 of July, 2004 [10:10:07] (:confused:) Hmm... Now I started to wonder, if it's somehow possible to override the canreinvite=no setting on per call basis. Anyone? H option by Anonymous on Saturday 10 of July, 2004 [04:15:13] Asterisk will not reinvite if the H option is used in the Dial command. http://www.voip-info.org/wiki-Asterisk+sip+canreinvite Thanks, Steve I understand. But because the majority of calls are not to be recorded, I don't have a need to keep Asterisk in the media path all the time. That's why I'm wondering if you could dynamically keep it in the media path or not. - Waldo On Dec 8, 2005, at 10:26 AM, Steve Totaro wrote: Well, then set canreinvite=no If that's the case, is it possible to override the canreinvite attribute of a SIP peer in extensions.conf before a call is made or answered by that peer? - Waldo On Dec 8, 2005, at 9:15 AM, Steve Totaro wrote: Is there a way to optionally keep asterisk in the media path in order to record calls using the Monitor command? For example, if I have a SIP peer that is defined with canreinvite=yes, this means that if possible, Asterisk will not be in the media path. However, what happens if the user presses something like *1 (defined in features.conf) to record the call? Will the call be forced to go through Asterisk automatically? Thanks, Waldo I could be wrong but I am pretty sure that once the asterisk is out of the media path then features like *1 will not work since asterisk is not able to listen for it. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Exit Voicemail
Voicemail in itself does not hangup, * will bring you back to the DP (to exten a). So if a user exits VM (I think they can exit by pressing # after recording) then you can drop them in a context that does what you want, you can do the same at exten a. On 12/8/05, Joe Pukepail [EMAIL PROTECTED] wrote: Is there a way to have control go back to the dialplan after a call gets to voicemail? I'm looking to implement findme and campon, but I want the options to be hidden, so if someone calling got a voicemail they could key in *1 (or whatever) and it would go back to the dialplan so I can implement fineme in the dial plan. The same with campon, if you got a busy voicemail you could key in *2 (or whatever) and it would take them to the piece of the dialplan where it would wait for person to get off the phone. I realize I could do this by having the user key in another option (Hit 1 to leave a voicemail, hit 2 to findme) but would prefer not to, users could record this as part of their voicemail message if they want the public to know about the findme and camping on a busy extension. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme and Sipura SPA-941 - bad jitter/distortion
I have a new * 1.2 server running on a dual-processor machine, 1GB of RAM, Gentoo with Linux 2.6 and a Digium TDM400 (four fxo boards) installed. Everything has been working great until we tried our first Meetme conference call yesterday. I have a total of 12 extensions. 9 of them are in the office with a direct connection to the server, all of the phones are Polycom 501s. The three remote users have the new Sipura SPA-941. I decided on this phone because of the features and it was easy to setup behind NAT (which all of these users have). Regular calls to these users work great with no issues at all. Its been wonderful. However, we had our first company conference via Meetme yesterday, and the SPA-941s sounded horrible in the conference. Very distorted, jittery sound. It was surprising and we ended up having them call in on the POTS line and come in that way and it sounded fine. So, I thought maybe it was a connection issue, but tested with one of our remote uses and have narrowed it down to the phone. If the user connects with X-lite to the conference room the sounds is great. If he then calls back with the SPA-941, the sound is horrible. Hanging up and dialing the extension directly to the SPA-941 sounds good as well. Any ideas what could be going on and how to fix it. I thought it could be a timing thing. The documentation on the Sipura phones is non-existent at the moment, so I have no idea what might be able to be changed. Id greatly appreciate any help or thoughts! Ryan Booz Director of IT Good Steward Software, LLC 111 Sowers Street, Suite 400 State College, PA 16801 Phone: 877-327-3702 x.26 (814-237-3744 x.26) Fax: 719-623-0577 Visit us at www.energycap.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Maximum Calls handled
If you have to ask this question, then the answer is, use Asterisk Business Edition. On 12/8/05, Goran Donev [EMAIL PROTECTED] wrote: I have a big dilemma. I have a client who is looking for a big installation. I am looking at the digium product and have the following Questions. Difference between Asterisk and Asterisk Business Edition. My Client has 300 personal split between two office and wants to use one asterisk box to support those calls. He is going to have 3 PRI'S coming in. Can I use the regular version of Asterisk compared to the Business Edition of Asterisk. How many simultaneous calls can Asterisk support compared to the Business Edition of Asterisk. Please help me out as I don't want to make the wrong recommendations. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Nortel Meridian Option81C to TE405P
We finally got the T1 connection working between the Nortel and the Asterisk box, but only with Robbed bit signalling. For some reason, the D channels would not come up when using PRI ISDN. No clue why, but I'm just happy to have it up and running. Anish Basu Field Systems Engineer Softel, Inc. Phone: (732) 705-9202 Cell: (732) 312-6634 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Aastra 9133i Configurations - are the file names to be lower case or upper case or does it matter?
On Wed, 2005-12-07 at 21:45 -0500, Lists wrote: According to the wiki page http://www.voip-info.org/tiki-index.php?page=Aastra+480i+Configuration it shows lowercase file name and then there is a comment at the bottom that it needs to be capitalized. I have tried it both ways with no luck. Could someone comment on which way the cfg files need to be in the /tftpboot directory? The letters in the MAC address have to be upper case, the extension lowercase. The best way to determine this is to put a -vv option on your tftp server (verbose) so you can see in the log file which file your phone is actually requesting. -- Carlos Chavez Director de Tecnologa Telecomunicaciones Abiertas de Mxico S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP.conf Technical Documentation - Help
Is there a document/wiki/web site that maps the various SIP.conf settings to the structure of the actual IP packet? If so please advise. -- ___ Play 100s of games for FREE! http://games.mail.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help iaxmodem
You can use app_nv_faxdetect. Hi: I want to use the same phone number for the fax and voice conversations. How do I redirect a call to the iaxmodem extension? Should my VOIP provider support the slinear codec? Thanks Miguel -Original Message- From: Miguel Soto [mailto:[EMAIL PROTECTED] Sent: Thursday, December 01, 2005 10:41 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] iaxmodem Hi: I want to use the same phone number for the fax and voice conversations. If it is a fax calling, I don't want any interactive menu, I just want to redirect the calling to the iaxmodem extension, and if is a normal calling the interactive menu will be deployed. How can I detect that is fax calling? Regards Miguel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Maximum Calls handled
I have a big dilemma. I have a client who is looking for a big installation. I am looking at the digium product and have the following Questions. Difference between Asterisk and Asterisk Business Edition. My Client has 300 personal split between two office and wants to use one asterisk box to support those calls. He is going to have 3 PRI'S coming in. Can I use the regular version of Asterisk compared to the Business Edition of Asterisk. How many simultaneous calls can Asterisk support compared to the Business Edition of Asterisk. Please help me out as I don't want to make the wrong recommendations. I may be totally wrong here but it is my understanding that ABE is basically the same as standard asterisk but you get a nice manual and support. http://www.digium.com/index.php?menu=product_detailcategory=softwarepr oduct=ABE You also get a little feel good to pass onto people that don't have faith in Open Source. Please correct me if I am wrong. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Call Recording and SIP canreinvite
Yeah, makes sense now that I think about it a little more. Guess you will have to prefix your exten so that the dial string with the H is used and dial that prefix when you know or think that you may have to record a call. This and Time Bandit's comment makes sense. I didn't realize that these options in the Dial string will force Asterisk to stay in the media path even if canreinvite=yes. I'll give it a try. Thanks, Waldo On Dec 8, 2005, at 11:18 AM, Steve Totaro wrote: There may be a better way but off the top of my head this idea jumped out. It assumes that you know prior to making the call that you need to record it and that you have phones capable of multiple lines. Setup a second line with a different entry in sip.conf with canreinvite=no and use that line to make your calls. Other than that I see reference on the wiki to an H option in dial but have never used it. I think you will still need to know prior to dialing whether you will want to record the call or not so you can dial the exten that uses the H option. If you get this to work, please post your results back to this thread. Re: Re: H option by flobi on Monday 25 of July, 2005 [10:43:46] why not just set canreinvite=yes and on the calls where you don't want reinvite use the H option (if it actually does disable reinvite) or the T or t which also disable reinvite. 7960G Seems to need canreinvite=no as well. by Anonymous on Friday 29 of October, 2004 [22:22:43] Running P0S3-07-2-00. Re: H option by Anonymous on Monday 26 of July, 2004 [10:10:07] (:confused:) Hmm... Now I started to wonder, if it's somehow possible to override the canreinvite=no setting on per call basis. Anyone? H option by Anonymous on Saturday 10 of July, 2004 [04:15:13] Asterisk will not reinvite if the H option is used in the Dial command. http://www.voip-info.org/wiki-Asterisk+sip+canreinvite Thanks, Steve I understand. But because the majority of calls are not to be recorded, I don't have a need to keep Asterisk in the media path all the time. That's why I'm wondering if you could dynamically keep it in the media path or not. - Waldo On Dec 8, 2005, at 10:26 AM, Steve Totaro wrote: Well, then set canreinvite=no If that's the case, is it possible to override the canreinvite attribute of a SIP peer in extensions.conf before a call is made or answered by that peer? - Waldo On Dec 8, 2005, at 9:15 AM, Steve Totaro wrote: Is there a way to optionally keep asterisk in the media path in order to record calls using the Monitor command? For example, if I have a SIP peer that is defined with canreinvite=yes, this means that if possible, Asterisk will not be in the media path. However, what happens if the user presses something like *1 (defined in features.conf) to record the call? Will the call be forced to go through Asterisk automatically? Thanks, Waldo I could be wrong but I am pretty sure that once the asterisk is out of the media path then features like *1 will not work since asterisk is not able to listen for it. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime Replication of a Single File
List users, Please provide me with tips on how to replicate a single file to a separate machine as changes are made to it. I would prefer a method that reacts to file modifications (ie. FAM/gamin) as opposed to timed loops/polling (cron + rsync). I'd also like to avoid NFS altogether. Keeping resource consumption low on the source machine is a priority. A bit of research has lead me to believe that calling rsync when gamin is alerted to a file modification would be a good fit for my scenario, but I'm unclear on the easiest implementation. My scenario is as follows. I have a machine that runs Asterisk VoIP PBX software. Asterisk creates a log file that we generate reports off of. Another machine handles the generation of these reports, which involves significant number crunching and file I/O. By replicating the file on the reporting machine, I'd like to decouple the resource consumption of reporting from the VoIP server. Some of the reports are used to monitor activities in realtime, so cronning off rsync on a large time interval is not an option. Thank you, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Meetme and Sipura SPA-941 - bad jitter/distortion
I'd greatly appreciate any help or thoughts! try: RTP Packet size on SIP tab ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme and Sipura SPA-941 - bad jitter/distortion
Any ideas what could be going on and how to fix it. I thought it could be a timing thing. The documentation on the Sipura phones is non-existent at the moment, so I have no idea what might be able to be changed. I’d greatly appreciate any help or thoughts! How about disabling silence suppression on the phones. Give it a try and see. -- Andres Technical Support http://www.telesip.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Meetme and Sipura SPA-941 - bad jitter/distortion
Are any of the phones setup to use a codec payload of more than 20ms? Bugid 5697 on the bug tracker has a patch to deal with very poor MeetMe performance when any of the participants are using audio packetization greater than 20ms. Beta1 and beta2 did not have this problem, and I am not sure about the RC versions. Which codec is the 941 using? Dan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ryan BoozSent: Thursday, December 08, 2005 8:27 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Meetme and Sipura SPA-941 - bad jitter/distortion I have a new * 1.2 server running on a dual-processor machine, 1GB of RAM, Gentoo with Linux 2.6 and a Digium TDM400 (four fxo boards) installed. Everything has been working great until we tried our first Meetme conference call yesterday. I have a total of 12 extensions. 9 of them are in the office with a direct connection to the server, all of the phones are Polycom 501s. The three remote users have the new Sipura SPA-941. I decided on this phone because of the features and it was easy to setup behind NAT (which all of these users have). Regular calls to these users work great with no issues at all. Its been wonderful. However, we had our first company conference via Meetme yesterday, and the SPA-941s sounded horrible in the conference. Very distorted, jittery sound. It was surprising and we ended up having them call in on the POTS line and come in that way and it sounded fine. So, I thought maybe it was a connection issue, but tested with one of our remote uses and have narrowed it down to the phone. If the user connects with X-lite to the conference room the sounds is great. If he then calls back with the SPA-941, the sound is horrible. Hanging up and dialing the extension directly to the SPA-941 sounds good as well. Any ideas what could be going on and how to fix it. I thought it could be a timing thing. The documentation on the Sipura phones is non-existent at the moment, so I have no idea what might be able to be changed. Id greatly appreciate any help or thoughts! Ryan Booz Director of IT Good Steward Software, LLC 111 Sowers Street, Suite 400 State College, PA 16801 Phone: 877-327-3702 x.26 (814-237-3744 x.26) Fax: 719-623-0577 Visit us at www.energycap.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OOH323 towards cisco gateway (2691) call setupfails at q931: Mandatory information element is missing (96)
Upgrade if you can. I remember submitting a report to the ooH323c developers about this some months ago and the fixed it right away. Dan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jacobso1Sent: Thursday, December 08, 2005 8:21 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] OOH323 towards cisco gateway (2691) call setupfails at q931: Mandatory information element is missing (96) Hi, I am using ooh323. I cannot setup a call towards a cisco gateway. The cisco rejects the call right away with : Cause value: Mandatory information element is missing (96) This is in the q931 part. Cisco explanation Indicates that the equipment that is sending this code has received a message that is missing an information element that must be present in the message before that message can be processed. Show version gives : Cvs-head-06/21/05-23:51:26 Someone any clue ? H323.conf : ; Objective System's H323 Configuration example for Asterisk ; ooh323c driver configuration ; ; [general] section defines global parameters ; ; This is followed by profiles which can be of three types - user/peer/friend ; Name of the user profile should match with the h323id of the user device. ; For peer/friend profiles, host ip address must be provided as "dynamic" is ; not supported as of now. ; ; Syntax for specifying a H323 device in extensions.conf is ; For Registered peers/friends profiles: ; H323/name where name is the name of the peer/friend profile. ; ; For unregistered H.323 phones: ; H323/ip[:port] OR if gk is used H323/alias where alias can be any H323 ; alias ; ; For dialing into another asterisk peer at a specific exten ; H323/exten/peer OR H323/[EMAIL PROTECTED] ; ; Domain name resolution is not yet supported. ; ; When a H.323 user calls into asterisk, his H323ID is matched with the profile ; name and context is determined to route the call ; ; The channel driver will register all global aliases and aliases defined in ; peer profiles with the gatekeeper, if one exists. So, that when someone ; outside our pbx (non-user) calls an extension, gatekeeper will route that ; call to our asterisk box, from where it will be routed as per dial plan. [general] ;Define the asetrisk server h323 endpoint ;The port asterisk should listen for incoming H323 connections. ;Default - 1720 port=1720 ;The dotted IP address asterisk should listen on for incoming H323 ;connections ;Default - tries to find out local ip address on it's own bindaddr=0.0.0.0 ;UPDATE this to proper ip address of your asterisk box ;Whether asterisk should use fast-start and tunneling for H323 connections. ;Default - yes faststart=yes h245tunneling=yes ;H323-ID to be used for asterisk server ;Default - Asterisk PBX h323id=TK_BRU_AST1 e164=100 ;CallerID to use for calls ;Default - Same as h323id callerid=TK_BRU_AST1 ;Whether this asterisk server will use gatekeeper. ;Default - DISABLE ;gatekeeper = DISCOVER ;gatekeeper = a.b.c.d gatekeeper = DISABLE ;Location for H323 log file ;Default - /var/log/asterisk/h323_log logfile=/var/log/asterisk/h323_log ;Following values apply to all users/peers/friends defined below, unless ;overridden within their client definition ;Sets default context all clients will be placed in. ;Default - default context=from-sip2 ;Sets rtptimeout for all clients, unless overridden ;Default - 60 seconds ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity ; when we're not on hold ;Type of Service ;Default - none (lowdelay, thoughput, reliability, mincost, none) ;tos=lowdelay ;amaflags = default ;The account code used by default for all clients. ;accountcode=h3230101 ;The codecs to be used for all clients. ;Default - ulaw ; ONLY ulaw, alaw, gsm, g729 and g723 (g723.1) are supported as of now disallow=all ;Note order of disallow/allow is important. allow=g729 allow=alaw allow=ulaw ; dtmf mode to be used by default for all clients. Only rfc2833 supported as ; of now. ;Default - rfc 2833 dtmfmode=rfc2833 ; User/peer/friend definitions: [TK_BRU_GW1] type=friend context=from-sip2 ip=195.xxx.yyy.zzz port=1720 disallow=all allow=g729 incominglimit=3 outgoinglimit=3 rtptimeout=60 dtmfmode=rfc2833 --No virus found in this outgoing message.Checked by AVG Free Edition.Version: 7.1.371 / Virus Database: 267.13.12/193 - Release Date: 6/12/2005 ___ --Bandwidth and
Re: [Asterisk-Users] Asterisk Call Recording and SIP canreinvite
I have a related issue. I have everything set up correctly so that I CAN use live recording (Press *1 to start and stop recording.) When I press *1, the console indicates user pressed *1 to start recording. I also hear the beep and an audio file is created. The problem is that the audio file IS NOTHING BUT SILENCE. It is the correct length, but only contains silence. Any ideas??? -N On Dec 8, 2005, at 8:49 AM, Steve Totaro wrote: Yeah, makes sense now that I think about it a little more. Guess you will have to prefix your exten so that the dial string with the H is used and dial that prefix when you know or think that you may have to record a call. This and Time Bandit's comment makes sense. I didn't realize that these options in the Dial string will force Asterisk to stay in the media path even if canreinvite=yes. I'll give it a try. Thanks, Waldo On Dec 8, 2005, at 11:18 AM, Steve Totaro wrote: There may be a better way but off the top of my head this idea jumped out. It assumes that you know prior to making the call that you need to record it and that you have phones capable of multiple lines. Setup a second line with a different entry in sip.conf with canreinvite=no and use that line to make your calls. Other than that I see reference on the wiki to an H option in dial but have never used it. I think you will still need to know prior to dialing whether you will want to record the call or not so you can dial the exten that uses the H option. If you get this to work, please post your results back to this thread. Re: Re: H option by flobi on Monday 25 of July, 2005 [10:43:46] why not just set canreinvite=yes and on the calls where you don't want reinvite use the H option (if it actually does disable reinvite) or the T or t which also disable reinvite. 7960G Seems to need canreinvite=no as well. by Anonymous on Friday 29 of October, 2004 [22:22:43] Running P0S3-07-2-00. Re: H option by Anonymous on Monday 26 of July, 2004 [10:10:07] (:confused:) Hmm... Now I started to wonder, if it's somehow possible to override the canreinvite=no setting on per call basis. Anyone? H option by Anonymous on Saturday 10 of July, 2004 [04:15:13] Asterisk will not reinvite if the H option is used in the Dial command. http://www.voip-info.org/wiki-Asterisk+sip+canreinvite Thanks, Steve I understand. But because the majority of calls are not to be recorded, I don't have a need to keep Asterisk in the media path all the time. That's why I'm wondering if you could dynamically keep it in the media path or not. - Waldo On Dec 8, 2005, at 10:26 AM, Steve Totaro wrote: Well, then set canreinvite=no If that's the case, is it possible to override the canreinvite attribute of a SIP peer in extensions.conf before a call is made or answered by that peer? - Waldo On Dec 8, 2005, at 9:15 AM, Steve Totaro wrote: Is there a way to optionally keep asterisk in the media path in order to record calls using the Monitor command? For example, if I have a SIP peer that is defined with canreinvite=yes, this means that if possible, Asterisk will not be in the media path. However, what happens if the user presses something like *1 (defined in features.conf) to record the call? Will the call be forced to go through Asterisk automatically? Thanks, Waldo I could be wrong but I am pretty sure that once the asterisk is out of the media path then features like *1 will not work since asterisk is not able to listen for it. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and
Re: [Asterisk-Users] Asterisk as a gatekeeper
Asterisk can only be configured as a GW AFAIK, what ever flavor of H323 you use with Asterisk it will not work as a GK. Atif rommel malana wrote: Hello, Right now i'm trying to set-up a gatekeeper and i'm having a hardtime doing it, what i'm thinking is instead of having a gatekeeper i'll use the asterisk to be a gatekeeper. Can the asterisk be a gatekeeper? Thanks a lot, Rommel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Octo Bri card together te405p and bristuff
On Thu, Dec 08, 2005 at 04:21:00PM +0100, Kib Eki wrote: Hi, is it possible to run an octoBri card together with a TE405P card in one system with bristuff? One question regarding zapbri: the bristuff patch is basically: 1. a major change to libpri 2. some relevant adjustments to chan_zap 3. other minor fixes and adjustments to asterisk 4. minor fixes to zaptel 5. three extra zaptel drivers To the octoBri card you basically need (5). Must some of the others be applied as well? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Lucent MAX TNT - how do I route a DID to my sip trunk
Currently Im running asterisk @ home 1.5 and a Lucent
Max TNT. I want to use the Max as a PSTN gateway for @home. To do
this I have a PRI terminated to the Max TNT.
As you can see below I have established a SIP trunk between
@home and the MAX TNT.
asterisk1*CLI sip show peers
Name/username
Host Dyn Nat
ACL
Mask
Port Status
maxtrunk1
172.16.255.191
255.255.255.255 5060 OK (15 ms)
230/230
172.16.255.200 D N
255.255.255.255 20924 Unmonitored
200/200
(Unspecified)
D 255.255.255.255
0 Unmonitored
asterisk1*CLI
From my softphone (ext. 230) I can dial out the Max TNT
successfully. I have setup a DID pointing to my softphone extension. E.G.
NPA-NXX-0230 - ext. 230.
Of course the DID terminates on the PRI connected to the Max
TNT. But when I call NPA-NXX-0230 from an outside PSTN line, I get this
message on the MAX.
LOG info, Shelf 1, Controller, Time: 14:40:28--
Releasing
[EMAIL PROTECTED]: Calling = NPANXX3405,Called =
NPANXX0230, Q850 Cause = 1,Sip Response = 404 (Not
Found),Progress Cause = NONE
LOG warning, Shelf 1, Slot 3, Time: 14:40:28--
[1/3/67/0] STOP: ''; cause 801.; progress 1404.; host
0.0.0.0 [MBID 71; NPANXX
3405-NPANXX0230]
I dont see any debug information come across my
terminal session with @home when I attempt to make the call.
What is necessary to make the Max TNT route the call to
@home when receiving a call for NPA-NXX-0230? And what do I need to do to route
100 DIDs to my @home box? Where in the Max do I put the range of
DIDs allocated to me and have the calls destined for them get passed
onto my @home box? Any help is greatly appreciated.
Marc
Below is most of the meat of my Max TNTs config.
[in MEDIA-GATEWAY/voip]
name* = voip
active = yes
protocol-type = sip
mgc-address = [ { 0.0.0.0 2944 } {
0.0.0.0 2944 } { 0.0.0.0 2944 } { +
mg-sig-address = { interface-dependent 0.0.0.0 }
mg-rtp-address = { system-default 0.0.0.0 }
h248-options = { text 3000 { no 0 } { 8000 6000 9000 [ {
} { } { +
ipdc-options = { IASCTNT1B { sig-queue-depth 60
send-info-to-mgc 120 reject-+
transport-options = { udp no { 0 1000 3000 3 7 6 } }
voip-options = { g711-ulaw { { yes 4 rtp yes } { yes 4
inband no } { no 1 rtp n+
dialed-gw-options = { disabled disabled disabled yes
ring-tone-on-alerting disa+
rt-fax-options = { no yes yes yes yes 0 no 14400 no }
tos-rtp-options = { no precedence-tos 00 000 normal }
tos-sig-options = { no precedence-tos 00 000 normal }
sip-options = { 500 4000 6 10 60 { 172.16.255.87
5060 compact { udp no { 0 0+
call-admission-control-options = { { yes } }
[in MEDIA-GATEWAY/voip:sip-options]
t1-timer = 500
t2-timer = 4000
invite-retries = 6
non-invite-retries = 10
tcp-idle-timer = 60
primary-proxy = { 172.16.255.87 5060 compact {
udp no { 0 0 0 0 0 0 } } }
secondary-proxy = { 0.0.0.0 5060 compact { udp
no { 0 0 0 0 0 0 } } }
registration-proxy = { 172.16.255.87 5060
compact { udp no { 0 0 0 0 0 0 } }+
proxy-heartbeat = 0
proxy-failover-window = 60
reroute-on-proxy-failure = no
trusted-proxy = { disabled [ { 0.0.0.0 } {
0.0.0.0 } { 0.0.0.0 } { +
unknown-ani = 00
unknown-name = www.rystec.com
blocked-ani = 00
blocked-name = blocked
privacy-proxy-require = disabled
isdn2sip-mapping = [ { 0 0 } { 0 0 } { 0 0 } { 0 0 } { 0 0 }
{ 0 0 } { 0 0 } { +
sip2isdn-mapping = [ { 0 0 } { 0 0 } { 0 0 } { 0 0 } { 0 0 }
{ 0 0 } { 0 0 } { +
start-call-method = invite
trunk-group-options = { prepend-to-userinfo no
prepend-to-userinfo }
onhold-minutes = 0
support-100rel = disabled
internationalize = no
international-prefix = no
country-code =
national-destination-code =
local-number-ton = unknown-ton
notify-timer = 0
options-trigger = [ { 488 304 } { 488 305 } { 606 304 } {
606 305 } { 415 304 }+
invite-with-multiple-codecs = disabled
egress-call-duration = 0
magic-number-prefix =
send-optional-headers = yes
user-agent-info = Lucent-Universal-Gateway
server-info = Lucent-Universal-Gateway
internationalize-cas = yes
T1/{ shelf-1 slot-2 1 } read
admin list
[in T1/{ shelf-1 slot-2 1 }]
name = ASTERISK-PRI-01
physical-address* = { shelf-1 slot-2 1 }
line-interface = { yes esf b8zs eligible middle-priority
isdn te wink-start dni+
autogenerated = no
[in T1/{ shelf-1 slot-2 1 }:line-interface]
enabled = yes
frame-type = esf
encoding = b8zs
clock-source = eligible
clock-priority = middle-priority
signaling-mode = isdn
isdn-emulation-side = te
robbed-bit-mode = wink-start
default-call-type = voip
switch-type = att-pri
nfas-group-id = 0
nfas-id = 0
incoming-call-handling = internal-processing
call-by-call = 0
network-specific-facilities = 0
data-sense = normal
idle-mode = flag-idle
FDL = none
front-end-type = dsx
DSX-line-length = 1-133
CSU-build-out = 0-db
overlap-receiving = no
pri-prefix-number =
tx-clir-flag-in-voip = no
trailing-digits = 2
t302-timer = 1
channel-config = [
[Asterisk-Users] Asterisk Bounty Pool
gNumber has written an application, UnWired Buyer, based on Asterisk. To show our thanks, we would like to extend an offer to the community. We are currently offering an PayPal credit of $10 to everyone that signs up and uses the service within the first 30 days. However if you use the promotion code of ASTERISK when you sign up, your $10 can be diverted into a Bounty Pool to pay for features in Asterisk. A committee of Asterisk developers is being assembled to accept projects and determine when they are complete. UnWired Buyer, our product based on Asterisk, is available at www.unwiredbuyer.com. It is an IVR that allows you to bid on auctions on eBay over a phone call. To earn the $10 for the Asterisk Bounty Pool you need to signup for UnWired Buyer and put ASTERISK in the Promotion Code field. Then within 30 days bid on some auction using UnWired Buyer. You are not required to win the auction but a bid must be placed using the system. For information on the program go to http://www.unwiredbuyer.com/asterisk -- Chris Tooley 512-646-1507 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Call Recording and SIP canreinvite
Hi! This and Time Bandit's comment makes sense. I didn't realize that these options in the Dial string will force Asterisk to stay in the media path even if canreinvite=yes. You might even have another option: DTMF via SIP INFO Quote from asterisk-devel two days ago: http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/14693 Cheers, Philipp Wolfgang S. Rupprecht wrote: I was thinking of hacking things a bit to allow my asterisk to stay out of the media path in the above case, but figured it couldn't hurt to post a quick sanity check here. Anyone see any problems? This is certainly possible, but Asterisk currently assumes that if it is not in the media path, it also won't be able to receive DTMF frames. However, if you are using SIP INFO for DTMF signaling, then it should 'just work', since when Asterisk sees the appropriate DTMF frames it will cause the bridge to 'break' and bring the media path back. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HDLC link unstable, yellow alarm on
This depends on the type of signalling you use! ISDN uses only 1 D channel that is chan 16 for EuroISDN. All other variants of ISDN (Q.Sig, 1TR, DPNSS, PSS1 etc) I know do the same. SS7/C7 is a different story as they can use up to 30 'links', but the most common is actually still 1 link at channel 16. jvb Andrew Latham wrote: If I remember correctly an E1 has two D-Channels. Check your notes on what channels 31, 32 really do. On 12/7/05, Laszlo Megyer [EMAIL PROTECTED] wrote: Hey folks, I have my linuxbox connected to a PBX through a digium te110p card, E1 line. The asterisk is set up to be the timing master for the line. recently run into the following error message when starting asterisk: The message: - == Primary D-Channel on span 1 down Dec 7 11:34:02 WARNING[1105]: chan_zap.c:2282 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! == Primary D-Channel on span 1 up Also the yellow alarm is not cleared on the PBX side. Any recommendations for the problem? Or should I use Tie-line connection between the PBX and asterisk somehow? thanks, lez config files follow: my /etc/zaptel.conf: ---8--- span=1,0,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 my /etc/asterisk/zapata.conf: 8--- [trunkgroups] [channels] spanmap = 1,1,1 usecallerid=yes hidecallerid=no language=en callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes callreturn=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no ; LOCAL CONFIGS: context = internal signalling = pri_net group = 1; channel = 1-15,17-31 switchtype = national ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP.conf Technical Documentation - Help
http://www.voip-info.org/wiki-asterisk http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf On 12/8/05, John Voss [EMAIL PROTECTED] wrote: Is there a document/wiki/web site that maps the various SIP.conf settings to the structure of the actual IP packet? If so please advise. -- ___ Play 100s of games for FREE! http://games.mail.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime Replication of a Single File
This sounds like a prime candidate for a database implementation. That way you can get very near real-time stats without the overhead of frequent cronjobs or polling. You number crunching computer would then just grab the data and crunch away. I'm just now getting started on using Asterisk in the more advanced modes (ie Realtime) so I do not know how to implement this, but I'm sure that it could be done. Ryan List users, Please provide me with tips on how to replicate a single file to a separate machine as changes are made to it. I would prefer a method that reacts to file modifications (ie. FAM/gamin) as opposed to timed loops/polling (cron + rsync). I'd also like to avoid NFS altogether. Keeping resource consumption low on the source machine is a priority. A bit of research has lead me to believe that calling rsync when gamin is alerted to a file modification would be a good fit for my scenario, but I'm unclear on the easiest implementation. My scenario is as follows. I have a machine that runs Asterisk VoIP PBX software. Asterisk creates a log file that we generate reports off of. Another machine handles the generation of these reports, which involves significant number crunching and file I/O. By replicating the file on the reporting machine, I'd like to decouple the resource consumption of reporting from the VoIP server. Some of the reports are used to monitor activities in realtime, so cronning off rsync on a large time interval is not an option. Thank you, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
