[asterisk-users] About the CALLIDNUMBER of the fxs
HI
Here is a question about the fxs of the zaptel card which is set a
number to use in the inter as common analog phone. When I also use
${CALLERID(num)}to get it's number, it also could not be done. At this time
,the fxs phone does not get any relation with the outbound which is like
PSTN and so forth. It just set the phone number and extension in the * for
inter used . Could you tell me the reason, and how I could get the number of
the fxs?
Thanks
Larry
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Re: [asterisk-users] Linksys - Sipura VMWI splash ring
On 08/21/08 21:10, Joseph wrote: I'm trying to configure Linksys 3102 for a short splash ring when someone leaves a message. in my sip.conf I have mailbox=number I have can see a visual indicator (light blinking on the phone) but there is no short splash ring) Linksys setting: Regional - tab Ring and Call Waiting Tone Spec Ring Waveform: Trapezoids Ring Frequency: 25 VMWI Refresh Intvl: 30 (was 0 I changed to 30 makes no difference) User 1 - tab Ring Settings: VMWI Ring Splash Len: 0.5 Did I miss any settings? Why isn't it working? It seems to me this feature is sync with Line 1 Register Expires: under: Proxy and Registration The default setting is 3600 so it means the phone will get a short ring every hour. If I want to ring the phone every 30sec I need to set Register Expires: 30 So I don't understand, what is the point of setting timer on: VMWI Refresh Intvl: since it doesn't get into effect until Register Expires I'm confused by this logic. - #Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] About the CALLIDNUMBER of the fxs
On Fri, 22 Aug 2008, larry wrote:
HI
Here is a question about the fxs of the zaptel card which is set a
number to use in the inter as common analog phone. When I also use
${CALLERID(num)}to get it's number, it also could not be done. At this
time ,the fxs phone does not get any relation with the outbound which is
like PSTN and so forth. It just set the phone number and extension in
the * for inter used . Could you tell me the reason, and how I could get
the number of the fxs?
Larry
As I understand your question...
You have a zaptel card with an analog phone connected to a FXS port.
You want CallerID associated with this line.
In your zapata.conf you should have the port defined kind of like:
context=local
signalling=fxo_ks
channel = 1
Just add the lines (before the channel callout!):
callerid=Common Phone Name common_phone_number_you_want
so it looks like:
context=local
signalling=fxo_ks
callerid=Larry's Phone 101
channel = 1
Ta Da!
Asterisk will use that as as your CallerID.
Brett
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[asterisk-users] set callerid with plus sign
Hi, Is it possible to assign a plus sign on the callerid(num) ? currently this is what i do CALLERID(num)=+6523450017 but telco is denying calls, coz they said they are seeing bs523450017 instead of +6523450017. i tried putting it inside double quotes CALLERID(num)=+6523450017 telco says the same thing. is this possible? thank you Regards, nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] set callerid with plus sign
Just change your dial command and add the plus sign there. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 22, 2008, at 1:28 AM, ronald wrote: Hi, Is it possible to assign a plus sign on the callerid(num) ? currently this is what i do CALLERID(num)=+6523450017 but telco is denying calls, coz they said they are seeing bs523450017 instead of +6523450017. i tried putting it inside double quotes CALLERID(num)=+6523450017 telco says the same thing. is this possible? thank you Regards, nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] interesting RDNIS question
Hi I am a premium voice service provider giving some services on IVR to a Telco X . As my premises is some 10 kms away from that telco , i have taken a PRI connection (30 DID with 1 hunting/pilot number) from telco Y When a customer of Telco X dials my short code @Rs.6/- per minute his call is forwarded on the PRI connection of telco Y . All this works fine.. Now the problem arises during billing , many customers of Telco X / Telco Z / Telco Y somehow get to know the pilot number of telco Y and they directly dial in (it becomes a local call and not a premium rate) the rsult being i dont get paid for those minutes and am giving the service free virtually ...I tried to solve the problem as follows : 1. If i filter the calls using DNIS - no matter people call short code or my pilot number - the DNIS would always be returned as the pilot number 2. If i filter calls using ANI so that i allow only customer of Telco X , then eventhough i minimise the damage - but still am not sure if that customer X has dialled short code or long code ? this question may sound off-topic but in asterisk is there a way out ? Rgds sriram___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] set callerid with plus sign
Hi Sir, I actually have a plus sign on my dial plan exten = _+.,1,Dial ( that is ok, dialed number (telco refers to it as B-number) is correct. the prob is the originating number(they call this A-Number), i want to set it to +65 so that it shows it is an international call. so on my dial plan: exten = _+.,1,Set(CALLERID(num)=+65) exten = _+.,1,Dial(SIP/[EMAIL PROTECTED]) what i don't get is why +65 is being seen as bs5. Regards, Nhadie Darren Sessions wrote: Just change your dial command and add the plus sign there. _ Darren Sessions [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 22, 2008, at 1:28 AM, ronald wrote: Hi, Is it possible to assign a plus sign on the callerid(num) ? currently this is what i do CALLERID(num)=+6523450017 but telco is denying calls, coz they said they are seeing bs523450017 instead of +6523450017. i tried putting it inside double quotes CALLERID(num)=+6523450017 telco says the same thing. is this possible? thank you Regards, nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemens Gigaset IP in USA (S685 IP in particular)
Another example of the North American frequency allocations being just a little bit different from everywhere else in the world... So does that mean you've stopped using your S685 IP, Michael? ;) Siemens USA does offer a few of the Gigaset DECT models over here (e.g. the E450, S450 and S455 are not hard to find). But none of the -IP models (e.g. no S450 IP). Perhaps they'd enjoy more success in the CE business if they offered those :) If WiFi handsets were more affordable I wouldn't have this problem :) Paul [EMAIL PROTECTED] wrote: That's the purely technological answer, which is completely correct. There's a business side to it as well. Siemens is simply not in the consumer electronics business in North America. They make this decision consciously. Michael Graves mgraves at mstvp.com o(713) 861-4005 c(713) 201-1262 sip:[EMAIL PROTECTED] skype mjgraves FWD 54245 Original Message Subject: Re: [asterisk-users] Siemens Gigaset IP in USA (S685 IP in particular) From: Drew Gibson [EMAIL PROTECTED] Date: Thu, August 21, 2008 12:08 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Paul Chambers wrote: For some unfathomable reason, Siemens USA doesn't offer the Gigaset IP range in the U.S. I'm particularly interested in the Gigaset S685 IP. Since it's DECT 6.0, and there's an English (UK) version, I'm thinking it should work just fine, after dealing with the walwart issue (and maybe caller ID signalling). Anyone imported one from the UK and using it in the US? for how long? impressions? anything not working? Have you purchased additional US-spec handsets and used them with the UK basestation? Thanks in advance, Paul The original DECT standard uses 1880-1900MHz, as implemented in Europe. The US FCC designated 1920-1930MHz. This is marketed as DECT 6.0. The FCC might get angry at you for using regular DECT phones in the US. And your neighbours with iPhones (GSM) might also get angry... regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with modem data calls and xorcom astribanks
On Fri, Aug 22, 2008 at 03:12:41PM +1000, Col Ferguson wrote: Hello all, I have a system at a motel that is mostly analog phones with 2 32 port astribanks. What exactly is the trunk? FXO ports in the astribank? I am having problems getting a modem data call to connect. There are many travelling salesmen that require this functionality to work to dial direct into their company systems. I am using Asterisk 1.4.18.1, and Zaptel 1.4.9.2 and freePBX 2.4.0.1 and Oslec echo can. I have now the simplest dialplan I can come up with and get a 4800 connection about 1 in 10 times. This should bypass any smarts that freePBX is adding in. The dialplan is [outbound-allroutes-custom] exten = 791,1,Dial(Zap/69/ww019830,300) exten = 791,n,Hangup In Hyperterminal I do atdt791 The number dialled is for a large dialup ISP. ww is needed to get a dialtone for the modem. Could this be causing the problem ? The log file shows [Aug 22 13:17:32] DEBUG[748] chan_zap.c: Deferring dialing... [Aug 22 13:17:32] VERBOSE[748] logger.c: -- Called 69/ww019830 [Aug 22 13:17:36] VERBOSE[748] logger.c: -- Zap/69-1 answered Zap/67-1 [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Took Zap/67-1 off hook [Aug 22 13:17:36] DEBUG[748] chan_zap.c: master: 67, slave: 69, nothingok: 0 [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Stopping tones on 67/0 talking to 69/0 [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Stopping tones on 69/0 talking to 67/0 [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Making 69 slave to master 67 at 0 [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Added 18 to conference 9/67 [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Added 76 to conference 9/69 [Aug 22 13:17:36] VERBOSE[748] logger.c: -- Native bridging Zap/67-1 and Zap/69-1 [Aug 22 13:20:00] DEBUG[748] chan_zap.c: Unlinking slave 69 from 67 [Aug 22 13:20:00] DEBUG[748] chan_zap.c: Removed 18 from conference 9/67 [Aug 22 13:20:00] DEBUG[748] chan_zap.c: Removed 76 from conference 9/69 [Aug 22 13:20:00] VERBOSE[748] logger.c: -- Hungup 'Zap/69-1' Does anyone know if there is some type of native echo canceller in the astribanks that could be affecting this ? Or anything else I could try ? Looking at /proc/oslec/info shows that oslec is not being used at the time. If I have the modem connected directly into the phone line, and completely bypass the astribank, I get a 50666 connection every time. Any suggestions gratefully accepted. Thanks, Col ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suddenly the voice become like robot (cutting), like sick man
Dear Darren; You might be right because one day it happened with me and the situation was same like this as following: The status that the ping result is very good for all partied (Asterisk machine, IP Phones on the Internet), and no problem in the processor utilization or RAM or hard disk space. Previously, we changed the DSL router and it worked fine !! But what can I do on the Asterisk level to overcome the problem? I already enabled the jitter on the IAX and SIP, but did not resolved. And I am using the G729 codec and sometimes I use GSM. Any advise for the robot voice with weak battery :) ?! Regards Bilal --- On Thu, 8/21/08, Darren Sessions [EMAIL PROTECTED] wrote: From: Darren Sessions [EMAIL PROTECTED] Subject: Re: [asterisk-users] Suddenly the voice become like robot (cutting), like sick man To: [EMAIL PROTECTED] Date: Thursday, August 21, 2008, 9:47 PM I doubt recompiling is going to help you unless you've got a very unstable system (hard drive going out or something), and then you've got bigger things to worry about then anyways. You should install (if you haven't already) the 'top' program. Top gives you a nice set of system statistics and a list of processes. If you're only having issues on the IP origination side of things, I would start checking your bandwidth and latency on your network. Is the originating end point on the Internet? or local? _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 21, 2008, at 4:55 PM, bilal ghayyad wrote: Dear Darren; I discovered that calling from the Asterisk to the IP Phone Extension (like calling from mobile to digium and then enter the IP Phone extension, or calling from fxs to the IP Phone extension), it goes very good without any problem. But calling from the same IP Phone to another IP Phone or to any mobile (via fxo port) or to the fxs, it cause the problem (voice become very very bad, like robot with weak battery or sick man). Another way for the problem, if I called from another Asterisk PBX to our Asterisk PBX (that has the problem) and the call was via IAX, and I was need to reach to the IP Phone, then I hear the voice like robot with weak battery. So, the problem appear if the call originator was IP and not TDM. What could be the reason for the problem? No one did any change, I am sure, it suddenly become like this. Any help? Regards Bilal --- On Thu, 8/21/08, Darren Sessions [EMAIL PROTECTED] wrote: From: Darren Sessions [EMAIL PROTECTED] Subject: Re: [asterisk-users] Suddenly the voice become like robot (cutting), like sick man To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non- Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, August 21, 2008, 6:13 PM I'd run top on the server to see if the CPU utilization is going through the roof. If you use AGI, make sure there aren't any orphaned processes consuming resources. If all else fails on the software side of things, I'd restart the server. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 21, 2008, at 4:03 PM, bilal ghayyad wrote: Hi All; My asterisk version is 1.4.19.2 and it contains one digium card of 2 fxs and 2 fxo ports, it was working great for more than one month without any problem. Suddenly, any call will be done, then voice becoming like robot (or sick man), it slow and cutting. I restarted the machine, but it is the same !!! I checked the RAM which is 1 GB and I found a lot of space. Any advise what could be the problem? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with modem data calls and xorcom astribanks
Hello Tzafrir, Yes the trunk is an FXO port in the astribank One astribank is 32 FXS ports, and one is 24 FXS and 8 FXO ports. Just in case it makes a difference, the testing I am doing is with the modem plugged in to the same astribank as the FXO ports. Zap/69 is an FXO port and Zap/67 is an FXS port Thanks, Col - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, August 22, 2008 6:56 PM Subject: Re: [asterisk-users] Problem with modem data calls and xorcom astribanks On Fri, Aug 22, 2008 at 03:12:41PM +1000, Col Ferguson wrote: Hello all, I have a system at a motel that is mostly analog phones with 2 32 port astribanks. What exactly is the trunk? FXO ports in the astribank? I am having problems getting a modem data call to connect. There are many travelling salesmen that require this functionality to work to dial direct into their company systems. I am using Asterisk 1.4.18.1, and Zaptel 1.4.9.2 and freePBX 2.4.0.1 and Oslec echo can. I have now the simplest dialplan I can come up with and get a 4800 connection about 1 in 10 times. This should bypass any smarts that freePBX is adding in. The dialplan is [outbound-allroutes-custom] exten = 791,1,Dial(Zap/69/ww019830,300) exten = 791,n,Hangup In Hyperterminal I do atdt791 The number dialled is for a large dialup ISP. ww is needed to get a dialtone for the modem. Could this be causing the problem ? The log file shows [Aug 22 13:17:32] DEBUG[748] chan_zap.c: Deferring dialing... [Aug 22 13:17:32] VERBOSE[748] logger.c: -- Called 69/ww019830 [Aug 22 13:17:36] VERBOSE[748] logger.c: -- Zap/69-1 answered Zap/67-1 [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Took Zap/67-1 off hook [Aug 22 13:17:36] DEBUG[748] chan_zap.c: master: 67, slave: 69, nothingok: 0 [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Stopping tones on 67/0 talking to 69/0 [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Stopping tones on 69/0 talking to 67/0 [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Making 69 slave to master 67 at 0 [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Added 18 to conference 9/67 [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Added 76 to conference 9/69 [Aug 22 13:17:36] VERBOSE[748] logger.c: -- Native bridging Zap/67-1 and Zap/69-1 [Aug 22 13:20:00] DEBUG[748] chan_zap.c: Unlinking slave 69 from 67 [Aug 22 13:20:00] DEBUG[748] chan_zap.c: Removed 18 from conference 9/67 [Aug 22 13:20:00] DEBUG[748] chan_zap.c: Removed 76 from conference 9/69 [Aug 22 13:20:00] VERBOSE[748] logger.c: -- Hungup 'Zap/69-1' Does anyone know if there is some type of native echo canceller in the astribanks that could be affecting this ? Or anything else I could try ? Looking at /proc/oslec/info shows that oslec is not being used at the time. If I have the modem connected directly into the phone line, and completely bypass the astribank, I get a 50666 connection every time. Any suggestions gratefully accepted. Thanks, Col ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] callfiles/manager api originate call fails
Actually both calls have to be originated to the outside world. Thats why im using @TRUNK-OUT, when the first call is answered only then the call goes to a context. Thats where the problem is, the first call does not originate so i cant throw it to any context. On Thu, Aug 21, 2008 at 8:47 PM, Anthony Francis [EMAIL PROTECTED]wrote: Rizwan Hisham wrote: Hi all, asterisk is giving me tough time. its been 3 days I am trying to originate outgoing call using manager api/callfiles. I would say remove the @TRUNK-OUT part and make sure that the context you send the call to knows about sending calls to the outside world. -- Thank you and have any kind of day you want, Anthony Francis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] set callerid with plus sign
Check on dial plan rules, remember if you need dial to +number your rule must be +|number this submit the number on your dialout plan without +. Regards, Luis Morales On Fri, Aug 22, 2008 at 3:40 AM, ronald [EMAIL PROTECTED] wrote: Hi Sir, I actually have a plus sign on my dial plan exten = _+.,1,Dial ( that is ok, dialed number (telco refers to it as B-number) is correct. the prob is the originating number(they call this A-Number), i want to set it to +65 so that it shows it is an international call. so on my dial plan: exten = _+.,1,Set(CALLERID(num)=+65) exten = _+.,1,Dial(SIP/[EMAIL PROTECTED]) what i don't get is why +65 is being seen as bs5. Regards, Nhadie Darren Sessions wrote: Just change your dial command and add the plus sign there. _ Darren Sessions [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 22, 2008, at 1:28 AM, ronald wrote: Hi, Is it possible to assign a plus sign on the callerid(num) ? currently this is what i do CALLERID(num)=+6523450017 but telco is denying calls, coz they said they are seeing bs523450017 instead of +6523450017. i tried putting it inside double quotes CALLERID(num)=+6523450017 telco says the same thing. is this possible? thank you Regards, nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing callerID in a context
On 21 Aug 2008, at 14:40, Philipp Kempgen wrote:
Andy Dixon schrieb:
I am trying to alter the outbound callerID for extensions within a
context I have created.
I wrote the following:
exten = _9.,2,ExecIf($[$[${REALCALLERIDNUM} = 360] | $[$
{REALCALLERIDNUM} = 670]]|Set|CALLERID(num)=581560)
exten = _9.,3,ExecIf($[$[${REALCALLERIDNUM} = 361] | $[$
{REALCALLERIDNUM} = 671]]|Set|CALLERID(num)=581561)
exten = _9.,4,ExecIf($[$[${REALCALLERIDNUM} = 362] | $[$
{REALCALLERIDNUM} = 672]]|Set|CALLERID(num)=581562)
exten = _9.,5,ExecIf($[$[${REALCALLERIDNUM} = 363] | $[$
{REALCALLERIDNUM} = 673]]|Set|CALLERID(num)=581563)
exten = _9.,6,ExecIf($[$[${REALCALLERIDNUM} = 364] | $[$
{REALCALLERIDNUM} = 674]]|Set|CALLERID(num)=581564)
exten = _9.,7,ExecIf($[$[${REALCALLERIDNUM} = 365] | $[$
{REALCALLERIDNUM} = 675]]|Set|CALLERID(num)=581565)
exten = _9.,8,ExecIf($[$[${REALCALLERIDNUM} = 366] | $[$
{REALCALLERIDNUM} = 676]]|Set|CALLERID(num)=581566)
exten = _9.,9,ExecIf($[$[${REALCALLERIDNUM} = 367] | $[$
{REALCALLERIDNUM} = 677]]|Set|CALLERID(num)=581567)
exten = _9.,10,ExecIf($[$[${REALCALLERIDNUM} = 368] | $[$
{REALCALLERIDNUM} = 678]]|Set|CALLERID(num)=581568)
exten = _9.,11,ExecIf($[$[${REALCALLERIDNUM} = 369] | $[$
{REALCALLERIDNUM} = 679]]|Set|CALLERID(num)=581569)
exten = _9.,12,ExecIf($[$[${REALCALLERIDNUM} = 700] | $[$
{REALCALLERIDNUM} = 701]]|Set|CALLERID(num)=581557)
exten = _9.,13,ExecIf($[$[${REALCALLERIDNUM} = 100] | $[$
{REALCALLERIDNUM} = 101]]|Set|CALLERID(num)=581500)
This *should* change the callerID for (for example) 700 and 701 to be
581557, and any extensions not listed above, it should leave them
alone.
If I call from extension 666, I get the correct outbound number (as
it
does exist), but the rules above are not being followed.
I have tried to use Set(CALLERID(num)=581500) which works okay
slightly further down.
I am aiming for any numbers starting with a 9 to follow the rules
above, and then to follow a further rule (eg if the number starts
901,
or 907)
I'm stuck.. If anyone could help, I would be eternally grateful..
This would be much more readable in AEL.
Or in an external script.
But maybe all you really need is fromuser in sip.conf or similar.
Hi,
I'm not wanting the callerID to change internally - eg so extension
700 will show as 700 internally, but as 581557 when it goes through
our ZAP trunk, which I believe the fromuser would cause.
Sorry to sound dim, but whats AEL?
Thanks
Andy
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Re: [asterisk-users] A Suggestion To Asterisk Appliance Developers
21 aug 2008 kl. 16.47 skrev [EMAIL PROTECTED] [EMAIL PROTECTED]: Yesterday I blogged a post about some ideas that I think will help Asterisk appliances further penetrate SMB/SOHO sites in ways that are not presently being addressed. I would prefer if you mailed the content too. After all this is a mailing list. Clicking on the link just to see what the topic is is something that most readers won't do. And it doesn't benefit the archives either. YOu can add a link, but just saying Hey, I blogged something interesting without saying anything about the topic is not very helpful. Just some friendly advice if you really want a discussion. Of course, I clicked, read and commented ;-) /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing callerID in a context
Andy Dixon wrote: On 21 Aug 2008, at 14:40, Philipp Kempgen wrote: This *should* change the callerID for (for example) 700 and 701 to be 581557, and any extensions not listed above, it should leave them alone. Andy, If you're not bound and determined to do it this way, you can use the MySQL addon to do a sql lookup for outbound CallerID. It works great. Just before the dial out, we do a query against the extension, if there is a match, we set the caller-id to whatever was returned on the query, if there isn't a match, we use the main number. If you'd like more information, let me know, Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a way to encrypt passwords stored in the realtime database?
Philippe Sultan [EMAIL PROTECTED] writes: Well, if someone steals the md5secret (HA1) for a given username and realm, he can use it to authenticate to the SIP proxy or B2BUA that serves the target user. This is unavoidable with password-based systems. Either you transfer the password unencrypted on the network (or e.g. hashed with MD5, but that just means that the hash is the actual password), and then you can store the password as a hash on the server. Or you use a secure protocol, e.g. a nonce-based one, to prove that the other end has the same password as you -- but then the server needs to have the unhashed password available for comparison. SIP tries to do both, but effectively it picks the second choice: Trust the server, not the network. To do better you need public key cryptography. Alas, noone has invented a way to create a private key from a password, so that means you don't get to pick your own private key. Still, I think that would have been a vastly better choice for SIP and for anything else where humans aren't expected to regularly type their password. Either way, SIP can't do it. You can also go the whole way with client certificates and SIP/TLS, and then you can hire a few people to keep your PKI running and secure -- and I'm not sure that Asterisk can do it yet. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk, Xen and a TDM400P
Hi, Has anybody managed to get this configuration work with PCI passthrough or should I look to buying a separate server ? Regards, -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: F57A 0CBD DD19 79E9 1FCC A612 CB36 D89D 2C5A 3A84 // Keyserver: www.keyserver.net Key-ID: 0x2C5A3A84 // Phone: +44 845 869 2749 SIP Phone: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to block incoming calls on PRI
On Thu, Aug 21, 2008 at 08:36:44PM -0500, Dwayne Hubbard wrote: I also want to reiterate that the libpri and Asterisk branches above are development branches, so be careful in a production environment. This functionality will be available in Asterisk 1.6.2. To disable a channel via the CLI type 'pri service disable channel chan' and to enable the channel type 'pri service enable channel chan'. That sounds cool. Two questions: 1) can you do it gracefully (both that and immediate are sometimes useful)? 2) can you take down either an entire span, or a channel range on the command line? (Oh yeah: can I backport it to 1.2? :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Huawei SoftX3000
Thanks for your answer and doing some test I have this SIP debug: From Huawei SIDE we have: 12:53:41.358166 IP (tos 0xb8, ttl 127, id 0, offset 0, flags [none], proto: UDP (17), length: 856) 189.8.113.170.5060 189.8.126.177.5060: SIP, length: 828 INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 189.8.113.170:5060;branch=z9hG4bKba4h2m2070fhnc4q20k1.1 Call-ID: [EMAIL PROTECTED] From: Anonymoussip:[EMAIL PROTECTED];tag=c8959281 To: sip:[EMAIL PROTECTED];user=phone CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:5060;transport=udp Supported: 100rel Privacy: user User-Agent: Huawei SoftX3000 V300R006 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,ME SSAGE,REFER Content-Length: 226 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 228436 228436 IN IP4 189.8.113.170 s=Sip Call c=IN IP4 189.8.113.170 t=0 0 m=audio 49606 RTP/AVP 8 0 97 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-15 and from Asterisk SIDE we have 12:54:30.364337 IP (tos 0x0, ttl 64, id 7173, offset 0, flags [none], proto: UDP (17), length: 791) 192.168.0.12.5060 189.8.113.170.5060: SIP, length: 763 SIP/2.0 200 OK Via: SIP/2.0/UDP 189.8.113.170:5060;branch=z9hG4bKba4h2m2070fhnc4q20k1.1;received=189.8.113.1 70 From: Anonymoussip:[EMAIL PROTECTED];tag=c8959281 To: sip:[EMAIL PROTECTED];user=phone;tag=as2daca8e1 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 259 v=0 o=root 4569 4569 IN IP4 192.168.0.12 s=session c=IN IP4 192.168.0.12 t=0 0 m=audio 11442 RTP/AVP 8 0 97 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 12:54:34.132505 IP (tos 0x0, ttl 64, id 7174, offset 0, flags [none], proto: UDP (17), length: 487) 192.168.0.12.5060 189.8.113.170.5060: SIP, length: 459 SIP/2.0 100 Trying Via: SIP/2.0/UDP 189.8.113.170:5060;branch=z9hG4bKo2k1gm10bgv1lcojk7c0.1;received=189.8.113.1 70 From: sip:[EMAIL PROTECTED];user=phone;tag=c963ca26 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Content-Length: 0 12:54:34.132729 IP (tos 0x0, ttl 64, id 7175, offset 0, flags [none], proto: UDP (17), length: 503) 192.168.0.12.5060 189.8.113.170.5060: SIP, length: 475 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 189.8.113.170:5060;branch=z9hG4bKo2k1gm10bgv1lcojk7c0.1;received=189.8.113.1 70 From: sip:[EMAIL PROTECTED];user=phone;tag=c963ca26 To: sip:[EMAIL PROTECTED];user=phone;tag=as3b751604 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Content-Length: 0 12:54:34.634264 IP (tos 0x0, ttl 64, id 7176, offset 0, flags [none], proto: UDP (17), length: 487) 192.168.0.12.5060 189.8.113.170.5060: SIP, length: 459 SIP/2.0 100 Trying Via: SIP/2.0/UDP 189.8.113.170:5060;branch=z9hG4bKo2k1gm10bgv1lcojk7c0.1;received=189.8.113.1 70 From: sip:[EMAIL PROTECTED];user=phone;tag=c963ca26 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Content-Length: 0 12:54:35.635851 IP (tos 0x0, ttl 64, id 7177, offset 0, flags [none], proto: UDP (17), length: 487) 192.168.0.12.5060 189.8.113.170.5060: SIP, length: 459 SIP/2.0 100 Trying Via: SIP/2.0/UDP 189.8.113.170:5060;branch=z9hG4bKo2k1gm10bgv1lcojk7c0.1;received=189.8.113.1 70 From: sip:[EMAIL PROTECTED];user=phone;tag=c963ca26 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces
Re: [asterisk-users] Changing callerID in a context
On Thu, Aug 21, 2008 at 3:11 PM, Andy Dixon [EMAIL PROTECTED] wrote:
Hello,
I am trying to alter the outbound callerID for extensions within a
context I have created.
I wrote the following:
exten = _9.,2,ExecIf($[$[${REALCALLERIDNUM} = 360] | $[$
{REALCALLERIDNUM} = 670]]|Set|CALLERID(num)=581560)
exten = _9.,3,ExecIf($[$[${REALCALLERIDNUM} = 361] | $[$
{REALCALLERIDNUM} = 671]]|Set|CALLERID(num)=581561)
exten = _9.,4,ExecIf($[$[${REALCALLERIDNUM} = 362] | $[$
{REALCALLERIDNUM} = 672]]|Set|CALLERID(num)=581562)
exten = _9.,5,ExecIf($[$[${REALCALLERIDNUM} = 363] | $[$
{REALCALLERIDNUM} = 673]]|Set|CALLERID(num)=581563)
exten = _9.,6,ExecIf($[$[${REALCALLERIDNUM} = 364] | $[$
{REALCALLERIDNUM} = 674]]|Set|CALLERID(num)=581564)
exten = _9.,7,ExecIf($[$[${REALCALLERIDNUM} = 365] | $[$
{REALCALLERIDNUM} = 675]]|Set|CALLERID(num)=581565)
exten = _9.,8,ExecIf($[$[${REALCALLERIDNUM} = 366] | $[$
{REALCALLERIDNUM} = 676]]|Set|CALLERID(num)=581566)
exten = _9.,9,ExecIf($[$[${REALCALLERIDNUM} = 367] | $[$
{REALCALLERIDNUM} = 677]]|Set|CALLERID(num)=581567)
exten = _9.,10,ExecIf($[$[${REALCALLERIDNUM} = 368] | $[$
{REALCALLERIDNUM} = 678]]|Set|CALLERID(num)=581568)
exten = _9.,11,ExecIf($[$[${REALCALLERIDNUM} = 369] | $[$
{REALCALLERIDNUM} = 679]]|Set|CALLERID(num)=581569)
exten = _9.,12,ExecIf($[$[${REALCALLERIDNUM} = 700] | $[$
{REALCALLERIDNUM} = 701]]|Set|CALLERID(num)=581557)
exten = _9.,13,ExecIf($[$[${REALCALLERIDNUM} = 100] | $[$
{REALCALLERIDNUM} = 101]]|Set|CALLERID(num)=581500)
This *should* change the callerID for (for example) 700 and 701 to be
581557, and any extensions not listed above, it should leave them alone.
If I call from extension 666, I get the correct outbound number (as it
does exist), but the rules above are not being followed.
I have tried to use Set(CALLERID(num)=581500) which works okay
slightly further down.
I am aiming for any numbers starting with a 9 to follow the rules
above, and then to follow a further rule (eg if the number starts 901,
or 907)
I'm stuck.. If anyone could help, I would be eternally grateful..
Are you sure ${REALCALLERIDNUM} is set? Alternatively (to AEL) there's
a way how to simplify all this, by using Asterisk extension patterns:
[clid-mangle]
exten = 70[01],1,Set(CALLERID(num)=581557)
exten = 70[01],2,Return()
exten = 10[01],1,Set(CALLERID(num)=581500)
exten = 10[01],2,Return()
; and so on, just better reorganize your extensions so that this can
match patterns better.
[dial-out]
exten = _9.,1,GoSub(clid-mangle,${CALLERID(num)},1)
exten = _9.,2,Dial(SIP/provider)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] set callerid with plus sign
ronald [EMAIL PROTECTED] writes: Is it possible to assign a plus sign on the callerid(num) ? Yes. currently this is what i do CALLERID(num)=+6523450017 but telco is denying calls, coz they said they are seeing bs523450017 instead of +6523450017. Which techology? SIP? PRI? POTS? ...? /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Asterisk-Stats - Billsec instead of Duration
On Thursday 21 August 2008 08:26:47 am Olivier wrote: Hi, To check telco billing, I'm usinfg Asterisk-Stats from http://www.areski.net/asterisk-stat-v2/about.php . How can you tweak this application to display graphics and data that use i started working from that software to come up that was maybe simpler and css-based. i'm still messing around with it, but you can look at https://messinet.com/svn/projects/asterisk-stat/trunk/ -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DSS1 vs SS7
Alex Balashov wrote: Some carriers now do offer private SS7 instead of ISDN. But there is absolutely no reason why you should be doing this with Asterisk. Asterisk-SS7 is quite tenuous at best. Unless you have some specific reason to be using it, don't. Actually, SS7 support in Asterisk 1.6.0 appears to be quite solid, and it is being used in a quite a number of production deployments. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemens Gigaset IP in USA (S685 IP in particular)
On Fri, 22 Aug 2008 01:13:25 -0700, Paul Chambers wrote: Another example of the North American frequency allocations being just a little bit different from everywhere else in the world... So does that mean you've stopped using your S685 IP, Michael? ;) Yes, between the power problems that I encountered (my own fault I suspect) and certain minor quirks of the phones I've gone back to my snom m3. Siemens USA does offer a few of the Gigaset DECT models over here (e.g. the E450, S450 and S455 are not hard to find). But none of the -IP models (e.g. no S450 IP). Perhaps they'd enjoy more success in the CE business if they offered those :) It's not at all claer what their thought process is about this. I know that VOIP Supply has tried to get access to their product line, but without success. If WiFi handsets were more affordable I wouldn't have this problem :) True enough. I was surprised at how well Polycom's SpectraLink 8002 worked with an appropriate access point. I still like the m3. Snom's commitment to continued firmware development gives me some hope that it's one or two shortcommings will eventually be overcome. Michael Paul [EMAIL PROTECTED] wrote: That's the purely technological answer, which is completely correct. There's a business side to it as well. Siemens is simply not in the consumer electronics business in North America. They make this decision consciously. Michael Graves mgraves at mstvp.com o(713) 861-4005 c(713) 201-1262 sip:[EMAIL PROTECTED] skype mjgraves FWD 54245 Original Message Subject: Re: [asterisk-users] Siemens Gigaset IP in USA (S685 IP in particular) From: Drew Gibson [EMAIL PROTECTED] Date: Thu, August 21, 2008 12:08 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Paul Chambers wrote: For some unfathomable reason, Siemens USA doesn't offer the Gigaset IP range in the U.S. I'm particularly interested in the Gigaset S685 IP. Since it's DECT 6.0, and there's an English (UK) version, I'm thinking it should work just fine, after dealing with the walwart issue (and maybe caller ID signalling). Anyone imported one from the UK and using it in the US? for how long? impressions? anything not working? Have you purchased additional US-spec handsets and used them with the UK basestation? Thanks in advance, Paul The original DECT standard uses 1880-1900MHz, as implemented in Europe. The US FCC designated 1920-1930MHz. This is marketed as DECT 6.0. The FCC might get angry at you for using regular DECT phones in the US. And your neighbours with iPhones (GSM) might also get angry... regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Internal Virus Database is out of date. Checked by AVG - http://www.avg.com Version: 8.0.138 / Virus Database: 270.6.4/1617 - Release Date: 8/17/2008 12:58 PM -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] set callerid with plus sign
+ is not a valid Caller*ID character. Asterisk allows you to use + in Caller*ID, but many carriers will reject the call if you do that. Benny Amorsen wrote: ronald [EMAIL PROTECTED] writes: Is it possible to assign a plus sign on the callerid(num) ? Yes. currently this is what i do CALLERID(num)=+6523450017 but telco is denying calls, coz they said they are seeing bs523450017 instead of +6523450017. Which techology? SIP? PRI? POTS? ...? /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing callerID in a context
Atis Lezdins wrote:
[clid-mangle]
exten = 70[01],1,Set(CALLERID(num)=581557)
exten = 70[01],2,Return()
exten = 10[01],1,Set(CALLERID(num)=581500)
exten = 10[01],2,Return()
; and so on, just better reorganize your extensions so that this can
match patterns better.
[dial-out]
exten = _9.,1,GoSub(clid-mangle,${CALLERID(num)},1)
exten = _9.,2,Dial(SIP/provider)
Actually, this can be even easier (although you didn't use the actual
CLID matching style):
[dial-out]
exten = _9./70[01],1,Set(CALLERID(num)=581557)
exten = _9./10[01],1,Set(CALLERID(num)=581500)
... more of the same 'priority 1' steps
exten = _9.,2,Dial(SIP/provider)
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - The Genuine Asterisk Experience (TM)
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Re: [asterisk-users] OT - Asterisk-Stats - Billsec instead of Duration
On Friday 22 August 2008 07:54:43 am Anthony Messina wrote: On Thursday 21 August 2008 08:26:47 am Olivier wrote: Hi, To check telco billing, I'm usinfg Asterisk-Stats from http://www.areski.net/asterisk-stat-v2/about.php . How can you tweak this application to display graphics and data that use i started working from that software to come up that was maybe simpler and css-based. i'm still messing around with it, but you can look at https://messinet.com/svn/projects/asterisk-stat/trunk/ sorry, i forgot to mention that all you'd need to change is the 'formatDuration' function in include/config.inc.php i currently have the billing duration contained in the abbr tag and it is displayed on hover over the duration. but that is easily switched. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suddenly the voice become like robot (cutting), like sick man
On Fri, Aug 22, 2008 at 5:14 AM, bilal ghayyad [EMAIL PROTECTED] wrote: Dear Darren; You might be right because one day it happened with me and the situation was same like this as following: The status that the ping result is very good for all partied (Asterisk machine, IP Phones on the Internet), and no problem in the processor utilization or RAM or hard disk space. Previously, we changed the DSL router and it worked fine !! But what can I do on the Asterisk level to overcome the problem? I already enabled the jitter on the IAX and SIP, but did not resolved. And I am using the G729 codec and sometimes I use GSM. Any advise for the robot voice with weak battery :) ?! Regards Bilal Try getting rid of IAX2 and use all SIP. This has fixed the issue you describe more than once in my experience. There are obviously many other things that can cause this, but IAX2 is the first to go when I am troubleshooting and there is no glaring reason why. You can usually setup OpenVPN for ease, or port forwarding to get around NAT issues. Give it a go, and let us know. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] interesting RDNIS question
I think you could minimize the incidence of the problem by having a PRI with like 100 numbers associated, with the CO doing the routing stripping off the last two forwarded digits. You also have a premium service provider that forwards premium calls to one of those numbers (I think from what you say that you do not see the premium number as the originating ID). You rotate the accepted number daily/weekly, so it's very unlikely that someone can simply guess it. Just my euro .02, l. In data Fri, 22 Aug 2008 09:37:20 +0200, Sriram [EMAIL PROTECTED] ha scritto: Hi I am a premium voice service provider giving some services on IVR to a Telco X . As my premises is some 10 kms away from that telco , i have taken a PRI connection (30 DID with 1 hunting/pilot number) from telco Y When a customer of Telco X dials my short code @Rs.6/- per minute his call is forwarded on the PRI connection of telco Y . All this works fine.. Now the problem arises during billing , many customers of Telco X / Telco Z / Telco Y somehow get to know the pilot number of telco Y and they directly dial in (it becomes a local call and not a premium rate) the rsult being i dont get paid for those minutes and am giving the service free virtually ...I tried to solve the problem as follows : 1. If i filter the calls using DNIS - no matter people call short code or my pilot number - the DNIS would always be returned as the pilot number 2. If i filter calls using ANI so that i allow only customer of Telco X , then eventhough i minimise the damage - but still am not sure if that customer X has dialled short code or long code ? this question may sound off-topic but in asterisk is there a way out ? Rgds sriram -- Home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A Suggestion To Asterisk Appliance Developers
On Fri, Aug 22, 2008 at 4:23 AM, Johansson Olle E [EMAIL PROTECTED] wrote: Just some friendly advice if you really want a discussion. Of course, I clicked, read and commented ;-) If this is a way we can get you to say something, Olle, I'm for it! :) This said, I think Michael was trying to work within the idea that those interested in this particular post might want to read it. Michael, if you (or anyone) want to post particular articles here, I'm sure everyone is ok with that. With the exception of spam, obviously. The problem too though is if there are images and links in the post, spam filters, etc. So in the long run, I'd say it's probably better just to do what he did :) Best, r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inefficient Codec Translation
Requesting help. Thaks On Tue, Aug 19, 2008 at 4:40 PM, Jim Boykin [EMAIL PROTECTED] wrote: We run asterisk to handle incoming DIDs and we have observed inefficient Codec Translation. Here is the scenario [DID Vendor] --- [Asterisk ] External GW [G729] | |--- External GW [iLBC] Our DID vendor and asterisk box supports both ilbc g729. However, our external gateway termination supports either ilbc or g729 (and not both) and depending on users location, we terminate it on either gateway. Since DID and asterisk box supports both the codecs, we assumed that asterisk will appropriately select codecs depending on where we terminate the call so that no codec translation happens. However, this seems to be an incorrect assumption and we see that different codecs get selected on two legs which leads to quality drop and extra CPU cycles. May be we are doing something wrong. Pls suggest what we are doing wrong. Below is asterisk configuration. [did] type=friend host=xxx canreinvite=yes disallow=all allow=g729 allow=ilbc [gw1] type=friend host=xxx canreinvite=yes disallow=all allow=g729 [gw2] type=friend host=xxx canreinvite=yes disallow=all allow=ilbc Thanks Jim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with modem data calls and xorcom astribanks
I have been told before on this list that a modem through a zaptel card will not work. And mine doesn't, at least not for data calls (it works fine for fax). Apparently the modem requires the full bandwidth of the POTS line, which you do not get through the zaptel card. You might at least check to make sure that echo cancellation is turned off. That can interfere with a data call. --Greg ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A Suggestion To Asterisk Appliance Developers
On 22 Aug 2008, at 14:55, randulo wrote: On Fri, Aug 22, 2008 at 4:23 AM, Johansson Olle E [EMAIL PROTECTED] wrote: Just some friendly advice if you really want a discussion. Of course, I clicked, read and commented ;-) If this is a way we can get you to say something, Olle, I'm for it! :) This said, I think Michael was trying to work within the idea that those interested in this particular post might want to read it. Michael, if you (or anyone) want to post particular articles here, I'm sure everyone is ok with that. With the exception of spam, obviously. The problem too though is if there are images and links in the post, spam filters, etc. So in the long run, I'd say it's probably better just to do what he did :) I'm more with Olle on this one. I often read this list offline (during my commute) and articles which reference a web page without at least summarizing the content are frustrating :-) Also the archives of this list are a valuable searchable resource (again available offline as spotlight indexes them for me on the my mac). So a short summary to accompany the link is great. Tim. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suddenly the voice become like robot (cutting), like sick man
It's tough to say why a voice would start sounding like a robot. There are so many variables that could effect your Asterisk server. I always go for process of elimination when I have a problem similar to this with call quality. What I would do is install an end point on the same local network / subnet as your asterisk server (either a hard phone or a soft phone like X-Lite by Counterpath). Register the phone locally with your Asterisk server and make some calls or put an echo tester up. If things sound good, you know your Asterisk server is working just fine, and the problems lies somewhere on your network between the Asterisk server and whatever gateway / device. If it sounds awful, and the codecs match, then it's time to start troubleshooting the server. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 22, 2008, at 3:14 AM, bilal ghayyad wrote: Dear Darren; You might be right because one day it happened with me and the situation was same like this as following: The status that the ping result is very good for all partied (Asterisk machine, IP Phones on the Internet), and no problem in the processor utilization or RAM or hard disk space. Previously, we changed the DSL router and it worked fine !! But what can I do on the Asterisk level to overcome the problem? I already enabled the jitter on the IAX and SIP, but did not resolved. And I am using the G729 codec and sometimes I use GSM. Any advise for the robot voice with weak battery :) ?! Regards Bilal --- On Thu, 8/21/08, Darren Sessions [EMAIL PROTECTED] wrote: From: Darren Sessions [EMAIL PROTECTED] Subject: Re: [asterisk-users] Suddenly the voice become like robot (cutting), like sick man To: [EMAIL PROTECTED] Date: Thursday, August 21, 2008, 9:47 PM I doubt recompiling is going to help you unless you've got a very unstable system (hard drive going out or something), and then you've got bigger things to worry about then anyways. You should install (if you haven't already) the 'top' program. Top gives you a nice set of system statistics and a list of processes. If you're only having issues on the IP origination side of things, I would start checking your bandwidth and latency on your network. Is the originating end point on the Internet? or local? _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 21, 2008, at 4:55 PM, bilal ghayyad wrote: Dear Darren; I discovered that calling from the Asterisk to the IP Phone Extension (like calling from mobile to digium and then enter the IP Phone extension, or calling from fxs to the IP Phone extension), it goes very good without any problem. But calling from the same IP Phone to another IP Phone or to any mobile (via fxo port) or to the fxs, it cause the problem (voice become very very bad, like robot with weak battery or sick man). Another way for the problem, if I called from another Asterisk PBX to our Asterisk PBX (that has the problem) and the call was via IAX, and I was need to reach to the IP Phone, then I hear the voice like robot with weak battery. So, the problem appear if the call originator was IP and not TDM. What could be the reason for the problem? No one did any change, I am sure, it suddenly become like this. Any help? Regards Bilal --- On Thu, 8/21/08, Darren Sessions [EMAIL PROTECTED] wrote: From: Darren Sessions [EMAIL PROTECTED] Subject: Re: [asterisk-users] Suddenly the voice become like robot (cutting), like sick man To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non- Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, August 21, 2008, 6:13 PM I'd run top on the server to see if the CPU utilization is going through the roof. If you use AGI, make sure there aren't any orphaned processes consuming resources. If all else fails on the software side of things, I'd restart the server. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 21, 2008, at 4:03 PM, bilal ghayyad wrote: Hi All; My asterisk version is 1.4.19.2 and it contains one digium card of 2 fxs and 2 fxo ports, it was working great for more than one month without any problem. Suddenly, any call will be done, then voice becoming like robot (or sick man), it slow and cutting. I restarted the machine, but it is the same !!! I checked the RAM which is 1 GB and I found a lot of space. Any advise what could be the problem? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] Problem with modem data calls and xorcom astribanks
Not sure what you've heard before, but I have successfully used a modem at 9600 baud (forced via AT commands) through a zaptel card on several occasions. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 22, 2008, at 8:14 AM, Greg Woods wrote: I have been told before on this list that a modem through a zaptel card will not work. And mine doesn't, at least not for data calls (it works fine for fax). Apparently the modem requires the full bandwidth of the POTS line, which you do not get through the zaptel card. You might at least check to make sure that echo cancellation is turned off. That can interfere with a data call. --Greg ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A Suggestion To Asterisk Appliance Developers
On Fri, Aug 22, 2008 at 7:18 AM, Tim Panton [EMAIL PROTECTED] wrote: I often read this list offline (during my commute) and articles which reference a web page without at least summarizing the content are frustrating :-) Not to argue, but to add that for what you describe I use Google Reader. I load up on bligs (such as Michael's) and read it all offline on trips. Of course Google Gears does not store images, but you have whatever part of the text is available via RSS. r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suddenly the voice become like robot (cutting), like sick man
Then after many hours of chasing ghosts, you decide, hmmm, let me eliminate IAX2 as a possible cause, and boom!, everything works and voice quality is perfect. Then you are happy you got it fixed but mad you you wasted so much time an the highly touted IAX2 that should just work but doesn't in many cases. Thanks, Steve T On Fri, Aug 22, 2008 at 10:22 AM, Darren Sessions [EMAIL PROTECTED] wrote: It's tough to say why a voice would start sounding like a robot. There are so many variables that could effect your Asterisk server. I always go for process of elimination when I have a problem similar to this with call quality. What I would do is install an end point on the same local network / subnet as your asterisk server (either a hard phone or a soft phone like X-Lite by Counterpath). Register the phone locally with your Asterisk server and make some calls or put an echo tester up. If things sound good, you know your Asterisk server is working just fine, and the problems lies somewhere on your network between the Asterisk server and whatever gateway / device. If it sounds awful, and the codecs match, then it's time to start troubleshooting the server. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 22, 2008, at 3:14 AM, bilal ghayyad wrote: Dear Darren; You might be right because one day it happened with me and the situation was same like this as following: The status that the ping result is very good for all partied (Asterisk machine, IP Phones on the Internet), and no problem in the processor utilization or RAM or hard disk space. Previously, we changed the DSL router and it worked fine !! But what can I do on the Asterisk level to overcome the problem? I already enabled the jitter on the IAX and SIP, but did not resolved. And I am using the G729 codec and sometimes I use GSM. Any advise for the robot voice with weak battery :) ?! Regards Bilal --- On Thu, 8/21/08, Darren Sessions [EMAIL PROTECTED] wrote: From: Darren Sessions [EMAIL PROTECTED] Subject: Re: [asterisk-users] Suddenly the voice become like robot (cutting), like sick man To: [EMAIL PROTECTED] Date: Thursday, August 21, 2008, 9:47 PM I doubt recompiling is going to help you unless you've got a very unstable system (hard drive going out or something), and then you've got bigger things to worry about then anyways. You should install (if you haven't already) the 'top' program. Top gives you a nice set of system statistics and a list of processes. If you're only having issues on the IP origination side of things, I would start checking your bandwidth and latency on your network. Is the originating end point on the Internet? or local? _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A Suggestion To Asterisk Appliance Developers
On Fri, 22 Aug 2008 13:23:09 +0200, Johansson Olle E wrote: 21 aug 2008 kl. 16.47 skrev [EMAIL PROTECTED] [EMAIL PROTECTED]: Yesterday I blogged a post about some ideas that I think will help Asterisk appliances further penetrate SMB/SOHO sites in ways that are not presently being addressed. I would prefer if you mailed the content too. After all this is a mailing list. Clicking on the link just to see what the topic is is something that most readers won't do. And it doesn't benefit the archives either. YOu can add a link, but just saying Hey, I blogged something interesting without saying anything about the topic is not very helpful. Just some friendly advice if you really want a discussion. Of course, I clicked, read and commented ;-) My appologies. I'm mindful of not posting inappropriately to mailing lists. Many thanks to those who read to, and the few who saw fit to comment. My premise is very simple. Any Asterisk Appliance in a small business stands a good change of being core infrastructure. If it has hooks to extend its reach easily into aspects of the business just beyond the basic telephony/UC sphere then it may be dramatically more valuable to the end user. In my particular case I have some nice Polycom and Aastra desk phones. I'd like to leverage the XHTML browsers in those phones to serve some utility functions, like opening an electric door release, electric gate, etc. I don't see why this sort of thing needs to be as difficult as it is presently. It seems that at the moment such matters are wholly DIY, or at best left to a consultant. This takes them out of the sphere of possibility of a large number of smaller installations, and therefore reduces the potential utility of the PBX. That seems a waste. The appliance approach is supposed to make things easier for end user sites. I think that we should take a broad view of that, and not focus solely on the telephony aspect. Consider the device a possible solution to a variety of business needs. Of course, there are limits. I'd never suggest a production Asterisk box be used as a file server beyond provisioning phones. Michael Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inefficient Codec Translation
On Tue, Aug 19, 2008 at 7:10 AM, Jim Boykin [EMAIL PROTECTED] wrote: We run asterisk to handle incoming DIDs and we have observed inefficient Codec Translation. Here is the scenario [DID Vendor] --- [Asterisk ] External GW [G729] | |--- External GW [iLBC] Our DID vendor and asterisk box supports both ilbc g729. However, our external gateway termination supports either ilbc or g729 (and not both) and depending on users location, we terminate it on either gateway. Since DID and asterisk box supports both the codecs, we assumed that asterisk will appropriately select codecs depending on where we terminate the call so that no codec translation happens. However, this seems to be an incorrect assumption and we see that different codecs get selected on two legs which leads to quality drop and extra CPU cycles. May be we are doing something wrong. Pls suggest what we are doing wrong. Below is asterisk configuration. [did] type=friend host=xxx canreinvite=yes disallow=all allow=g729 allow=ilbc [gw1] type=friend host=xxx canreinvite=yes disallow=all allow=g729 [gw2] type=friend host=xxx canreinvite=yes disallow=all allow=ilbc Thanks Jim Why don't you allow=g729 only on all entries. Maybe I have misread your email but I interpret what you wrote to mean that all endpoints support g729 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inefficient Codec Translation
On Tue, Aug 19, 2008 at 7:10 AM, Jim Boykin [EMAIL PROTECTED] wrote: We run asterisk to handle incoming DIDs and we have observed inefficient Codec Translation. Here is the scenario [DID Vendor] --- [Asterisk ] External GW [G729] | |--- External GW [iLBC] Our DID vendor and asterisk box supports both ilbc g729. However, our external gateway termination supports either ilbc or g729 (and not both) and depending on users location, we terminate it on either gateway. Since DID and asterisk box supports both the codecs, we assumed that asterisk will appropriately select codecs depending on where we terminate the call so that no codec translation happens. However, this seems to be an incorrect assumption and we see that different codecs get selected on two legs which leads to quality drop and extra CPU cycles. May be we are doing something wrong. Pls suggest what we are doing wrong. Below is asterisk configuration. [did] type=friend host=xxx canreinvite=yes disallow=all allow=g729 allow=ilbc [gw1] type=friend host=xxx canreinvite=yes disallow=all allow=g729 [gw2] type=friend host=xxx canreinvite=yes disallow=all allow=ilbc Thanks Jim To be more clear, when a call comes in on [did] the codec should match on the order that they are listed and supported, so you should always get g729 and transcode to ilbc on gw2. Asterisk handles one leg at a time, so it does not look ahead to see what the second leg of the call will be using for it's codec. Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DSS1 vs SS7
Kevin P. Fleming wrote: Alex Balashov wrote: Some carriers now do offer private SS7 instead of ISDN. But there is absolutely no reason why you should be doing this with Asterisk. Asterisk-SS7 is quite tenuous at best. Unless you have some specific reason to be using it, don't. Actually, SS7 support in Asterisk 1.6.0 appears to be quite solid, and it is being used in a quite a number of production deployments. Thanks for the plug Kevin! :-) Yeah, actually, if you guys want to know more there's an asterisk-ss7 mailing list. Asterisk-1.6.0 with libss7 is being used in many successful and high traffic installations around the world. The current record (that I have been told of) is an installation doing over 100,000 calls per day. So try to beat that ;-) Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Diamondware spatial conferencing
Just read this on Alec Saunders fantastic blog http://saunderslog.com/2008/08/21/diamondware-acquired/ Interesting concept, makes me wonder if it is possible in Asterisk to 'adjust' the left/right mix for audio conference participants? Yeh I know there is only one channel in a telephone call but you get what I mean. It's probably covered under patents etc but has anyone tried to 'spatially separate' audio mixes for each participant in an Asterisk conference call? Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 (New York) +61-2-9016-5642 (Sydney) http://www.Cognation.net http://www.Cognation.net/profile ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday's conference meeting - Astricon is in the air
We will be gathering at 9AM PDT, 12 Noon EDT, 4PM GMT (i think?) for our weekly conference of asterisk and VoIP users. Any and all discussion related to telephony is welcome. Please join us any Friday. More info using the links below. PSTN (724) 444-7444 and enter 22622# 1# SIP [EMAIL PROTECTED] DTMF 22622# 1# IRC.freenode.net #voip-users-conference See http://bit.ly/voip for more info. Note: we were supposed to hold the drawing for the free Astricon pass, but Stelios was not able to do this today. Hopefully next week. Randy http://voipUsersConference.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diamondware spatial conferencing
On Fri, Aug 22, 2008 at 7:55 AM, Dean Collins [EMAIL PROTECTED] wrote: It's probably covered under patents etc but has anyone tried to 'spatially separate' audio mixes for each participant in an Asterisk conference call? I've never tried it, but as soon as I heard about the idea a few months ago, I thought it sounded like a great improvement to the presence you feel in a conference. r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RES: DSS1 vs SS7
Hello All, I have an Asterisk Box currently running 1.6.0, dahdi drivers and libss7 with a TE120P (01 E1) card for the past 02 months. I have it connected to a Cellular Operator switch (MSC), and it is working perfectly. Traffic is still quite low, but increasing as we start to use it for new applications everyday. I have made some stress call tests, using all available CICs at once, and had no problem at all. Congrats to the perfect development of the SS7 support to Mr. Fredrickson. Hopefully soon we'll have MAP support as well. Marco -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Matthew Fredrickson Enviada em: sexta-feira, 22 de agosto de 2008 11:49 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] DSS1 vs SS7 Kevin P. Fleming wrote: Alex Balashov wrote: Some carriers now do offer private SS7 instead of ISDN. But there is absolutely no reason why you should be doing this with Asterisk. Asterisk-SS7 is quite tenuous at best. Unless you have some specific reason to be using it, don't. Actually, SS7 support in Asterisk 1.6.0 appears to be quite solid, and it is being used in a quite a number of production deployments. Thanks for the plug Kevin! :-) Yeah, actually, if you guys want to know more there's an asterisk-ss7 mailing list. Asterisk-1.6.0 with libss7 is being used in many successful and high traffic installations around the world. The current record (that I have been told of) is an installation doing over 100,000 calls per day. So try to beat that ;-) Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diamondware spatial conferencing
This is a poor example but basically if you have 'stereo' I was thinking an equation like this. Number of speakers in a conference room = 'N' deviations Range = 80% (obviously you couldn't do 100% otherwise would be silent in outer channels once you get over 10 participants. (50/50 + / - the deviation N / range) So in the example of a conference room with 3 participants deviation 26% Caller 1 = 76% L + 24% R(-1 deviation of 26%) Caller 2 = 50% L + 50% R(norm default 50/50) Caller 3 = 24% L + 76% R(+1 deviation of 26%) Example of a conference room with 7 participants deviation of 11% Caller 1 = 83% L + 17% R(-3 deviations of 33%) Caller 2 = 72% L + 28% R(-2 deviations of 22%) Caller 3 = 61% L + 39% R(-1 deviations of 11%) Caller 4 = 50% L + 50% R(norm default 50/50) Caller 5 = 39% L + 61% R(+1 deviations of 11%) Caller 6 = 28% L + 72% R(+2 deviations of 22%) Caller 7 = 17% L + 83% R(+3 deviations of 33%) Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of randulo Sent: Friday, 22 August 2008 11:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Diamondware spatial conferencing On Fri, Aug 22, 2008 at 7:55 AM, Dean Collins [EMAIL PROTECTED] wrote: It's probably covered under patents etc but has anyone tried to 'spatially separate' audio mixes for each participant in an Asterisk conference call? I've never tried it, but as soon as I heard about the idea a few months ago, I thought it sounded like a great improvement to the presence you feel in a conference. r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [SOLVED] Linksys SPA3102-NA firmware upgrade on Linux
On 08/20/08 14:46, Paul Hales wrote: Joseph wrote: Does anybody know if the process of upgrading firmware on Linksys SPA3102-NA in Linux is the same as on Sipura 3K as described on voip-info.org http://www.voip-info.org/wiki/view/Sipura I'm pretty sure it works - I used it to upgrade a (god help me) SPA 9000 the other week. PaulH No it DID NOT work for me - upgrading via http web-server as described in: http://www.voip-info.org/wiki/index.php?page=Linksys-Cisco+3102 under: Firmware section So I was forced myself to use an EXE file (what a pity). However, I got in interesting post from a guru on Voxilla forum: quote Firmware can be loaded by placing a URL into Upgrade Rule (under Provisioning tab). Example: Place the following URL into the Upgrade Rule space: http://www.linuxstation.net/pub/voip/Linksys/Firmware/spa3102-5-1-7.bin then save the changes. The ATA will reboot, then it will load the new firmware and reboot again. -end quote-- I'll try it the next time, and I think www.voip-info.org should be appended with this information as good information is hard to find. -- #Joseph GPG KeyID: ED0E1FB7 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to block incoming calls on PRI
- Jay R. Ashworth [EMAIL PROTECTED] wrote: 1) can you do it gracefully (both that and immediate are sometimes useful)? Right now you can only disable an idle channel. 2) can you take down either an entire span, or a channel range on the command line? This functionality will be added after the current development branches are merged into trunk. I'm capping the feature creep on these branches so we can get this functionality tested and committed. (Oh yeah: can I backport it to 1.2? :-) You probably could but I don't know how cleanly the patches will apply. -Dwayne. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queue timeout
Hello,
I want to ask, how to detect queue timeout? If queue members are busy or
not answering to the call, and after queue timeout caller would hear :
Sorry all operators are busy, please leave a record:
This example:
[ivr]
exten = start,1,Ringing
exten = start,n,Wait(2)
exten = start,n,Answer
exten = start,n,Playback(ivr/welcome)
exten =
start,n,Set(RECORD_FILENAME=/var/spool/asterisk/monitor/ivr/operator-${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}-${CALLERID(num)})
exten = start,n,Set(MONITOR_FILENAME=${RECORD_FILENAME})
exten = start,n,Queue(ivr|tT|||30)
exten = t,1,Goto(ivr,recording,1)
exten = recording,1,Playback(ivr/leave-the-message)
exten =
recording,n,Set(RECORD_FILENAME=/var/spool/asterisk/monitor/ivr/irasas-${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}-${CALLERID(num)})
exten = recording,n,Record(${RECORD_FILENAME}:wav||60)
So if operators are busy or not answering, and after 30 sec, I want to run
recording. But if operators answered the call, I want just hangup call. And
I don't no how to do that. Maybe use Dial command instead using Queue ? Or
is another way
Thanks for your help.
--
Pagarbiai / Best Regards,
Giedrius Augys
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[asterisk-users] em wink
Are there parameters for em wink? 1) timing parameters 2) dial delay or pre dial. Thanks Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue timeout
Giedrius Augys wrote:
Hello,
I want to ask, how to detect queue timeout? If queue members are
busy or not answering to the call, and after queue timeout caller would
hear : Sorry all operators are busy, please leave a record:
This example:
[ivr]
exten = start,1,Ringing
exten = start,n,Wait(2)
exten = start,n,Answer
exten = start,n,Playback(ivr/welcome)
exten =
start,n,Set(RECORD_FILENAME=/var/spool/asterisk/monitor/ivr/operator-${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}-${CALLERID(num)})
exten = start,n,Set(MONITOR_FILENAME=${RECORD_FILENAME})
exten = start,n,Queue(ivr|tT|||30)
exten = t,1,Goto(ivr,recording,1)
exten = recording,1,Playback(ivr/leave-the-message)
exten =
recording,n,Set(RECORD_FILENAME=/var/spool/asterisk/monitor/ivr/irasas-${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}-${CALLERID(num)})
exten = recording,n,Record(${RECORD_FILENAME}:wav||60)
So if operators are busy or not answering, and after 30 sec, I want to
run recording. But if operators answered the call, I want just hangup
call. And I don't no how to do that. Maybe use Dial command instead
using Queue ? Or is another way
Thanks for your help.
--
Pagarbiai / Best Regards,
Giedrius Augys
You can check the value of QUEUESTATUS after the call to Queue(). If the call
timed out, then it will be set to TIMEOUT.
Mark Michelson
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Re: [asterisk-users] em wink
On Fri, 2008-08-22 at 14:40 -0400, Jerry Geis wrote: Are there parameters for em wink? A quick glance at the sample zapata.conf that comes with Asterisk shows prewink, wink, and rxwink timing parameters. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue timeout
On Fri, 2008-08-22 at 21:26 +0300, Giedrius Augys wrote: I want to ask, how to detect queue timeout? If queue members are busy or not answering to the call, and after queue timeout caller would hear : Sorry all operators are busy, please leave a record: The Queue() application sets a channel variable named QUEUESTATUS, and by reading that channel variable, you'll be able to evaluate why the call left the queue (due to a timeout or other reason). See the end of the description for the queue application in the online help by typing core show application queue at the Asterisk command-line interface. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue timeout
On Aug 22, 2008, at 2:26 PM, Giedrius Augys wrote:
Hello,
I want to ask, how to detect queue timeout? If queue members are
busy or not answering to the call, and after queue timeout caller
would hear : Sorry all operators are busy, please leave a record:
This example:
[ivr]
exten = start,1,Ringing
exten = start,n,Wait(2)
exten = start,n,Answer
exten = start,n,Playback(ivr/welcome)
exten = start,n,Set(RECORD_FILENAME=/var/spool/asterisk/monitor/ivr/
operator-${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}-${CALLERID(num)})
exten = start,n,Set(MONITOR_FILENAME=${RECORD_FILENAME})
exten = start,n,Queue(ivr|tT|||30)
exten = start,n,voicemail(ACCOUNT)
Fred Posner
www.teamforrest.com
smime.p7s
Description: S/MIME cryptographic signature
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Re: [asterisk-users] About the CALLIDNUMBER of the fxs
larry schrieb: Here is a question about the fxs of the zaptel card which is set a Didn't you post almost the exact same question yesterday? (twice already) Grüße, Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL (was: Re: Changing callerID in a context)
Andy Dixon schrieb: whats AEL? Asterisk Extension Language. extensions.ael Grüße, Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inefficient Codec Translation
Steve Totaro wrote: On Tue, Aug 19, 2008 at 7:10 AM, Jim Boykin [EMAIL PROTECTED] wrote: We run asterisk to handle incoming DIDs and we have observed inefficient Codec Translation. Here is the scenario [DID Vendor] --- [Asterisk ] External GW [G729] | |--- External GW [iLBC] Our DID vendor and asterisk box supports both ilbc g729. However, our external gateway termination supports either ilbc or g729 (and not both) and depending on users location, we terminate it on either gateway. Since DID and asterisk box supports both the codecs, we assumed that asterisk will appropriately select codecs depending on where we terminate the call so that no codec translation happens. However, this seems to be an incorrect assumption and we see that different codecs get selected on two legs which leads to quality drop and extra CPU cycles. May be we are doing something wrong. Pls suggest what we are doing wrong. Below is asterisk configuration. [did] type=friend host=xxx canreinvite=yes disallow=all allow=g729 allow=ilbc [gw1] type=friend host=xxx canreinvite=yes disallow=all allow=g729 [gw2] type=friend host=xxx canreinvite=yes disallow=all allow=ilbc Thanks Jim Why don't you allow=g729 only on all entries. Maybe I have misread your email but I interpret what you wrote to mean that all endpoints support g729 I may be wrong but I understood the situation as the DID supplier supports either g.729 or ilibc, but the user has 2 locations that calls are routed to. One location supports iLibc only, the other supports g.729 only. What they seem to be trying to accomplish is to get the DID - Asterisk leg to use the same codec as the Asterisk - Remote Location leg. I think the problem is going to be that the call has to be established to the Asterisk box before a destination can be selected. The DID and Asterisk Box are going to negotiate the first available common codec before doing anything else, including setting a destination. Since you can't change a codec once a call has been established you're always going to end up with calls to one of the 2 remote locations being transcoded. The only solution I could think of would be if there was some way to identify which incoming calls were going to be routed to which location and set the codec accordingly. To do that, you'd either have to have 2 different DID's or some other massively more complicated mechanism. Forcing a reinvite (Is that even possible?) would be the only other long-shot I could think of. Good luck, Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fw: [asterisk-dev] frequent channel reset problem
Sorry to have posted to the wrong maillist. Repost here. Regards. --- On Fri, 8/22/08, Ming-Ching Tiew [EMAIL PROTECTED] wrote: From: Ming-Ching Tiew [EMAIL PROTECTED] Subject: [asterisk-dev] frequent channel reset problem To: [EMAIL PROTECTED] Date: Friday, August 22, 2008, 3:04 PM Hi, I am stucked with a nasty PRI problem for 2 weeks now and will appreciate if I could get some help from here. The problem is that my zaptel.conf and zapata.conf have been working with one PRI line but it is not working with another PRI. I could easily blame it to the PRI line quality itself, but the fact is that the PRI line has been working fine with at least 2 PBXes for ages without experiencing channel reset or call drop problem. However when connected to asterisk using wct4xxp driver, it will get call drop randomly every now and then. From the asterisk console output, when calls are dropped, there is associated alarm detected. I would like to know the following: (1) After detecting an alarm, does the ISDN spec. specifies that all channels must be resetted or this is just an implementation choice which asterisk has made ? Should asterisk reset the line after detecting the alarm and caused all calls being drop? (2) I have a pri monitor ( ISDN tester) that sniffs the PRI line, I found that whenever the dropping calls problem happens, the monitor print out RDI Begin ... SAMBE ,RESTART, RESTART ACK... RDI end. Does anyone know what is RDI other than what stated in the monitor's manual as Remote Defect Detect and what causes it. Attached is the PRI intense debug file. ( http://www.geocities.com/mctiew/asterisk/bd2.txt ) Any pointer or help given is greatly appreciated ! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fw: [asterisk-dev] frequent channel reset problem
From: Ming-Ching Tiew [EMAIL PROTECTED] Subject: [asterisk-dev] frequent channel reset problem To: [EMAIL PROTECTED] Date: Friday, August 22, 2008, 3:04 PM Hi, I am stucked with a nasty PRI problem for 2 weeks now and will appreciate if I could get some help from here. The problem is that my zaptel.conf and zapata.conf have been working with one PRI line but it is not working with another PRI. I could easily blame it to the PRI line quality itself, but the fact is that the PRI line has been working fine with at least 2 PBXes for ages without experiencing channel reset or call drop problem. However when connected to asterisk using wct4xxp driver, it will get call drop randomly every now and then. From the asterisk console output, when calls are dropped, there is associated alarm detected. I would like to know the following: (1) After detecting an alarm, does the ISDN spec. specifies that all channels must be resetted or this is just an implementation choice which asterisk has made ? Should asterisk reset the line after detecting the alarm and caused all calls being drop? (2) I have a pri monitor ( ISDN tester) that sniffs the PRI line, I found that whenever the dropping calls problem happens, the monitor print out RDI Begin ... SAMBE ,RESTART, RESTART ACK... RDI end. Does anyone know what is RDI other than what stated in the monitor's manual as Remote Defect Detect and what causes it. Attached is the PRI intense debug file. ( http://www.geocities.com/mctiew/asterisk/bd2.txt ) Any pointer or help given is greatly appreciated ! Try setting resetinterval to never. resetinterval: sets the time in seconds between restart of unused channels, defaults to 3600 minimum 60 seconds. Some PBXs don't like channel restarts. so set the interval to a very long interval e.g. 1 or 'never' to disable *entirely*. http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] frequent channel reset problem
Sorry to have posted reply to your email directly. Repost to maillist. resetinterval already set to never to start with. Thanks --- On Fri, 8/22/08, Ming-Ching Tiew [EMAIL PROTECTED] wrote: From: Ming-Ching Tiew [EMAIL PROTECTED] Subject: Re: [asterisk-users] Fw: [asterisk-dev] frequent channel reset problem To: Steve Totaro [EMAIL PROTECTED] Date: Friday, August 22, 2008, 10:58 PM --- On Fri, 8/22/08, Steve Totaro [EMAIL PROTECTED] wrote: From: Steve Totaro [EMAIL PROTECTED] Try setting resetinterval to never. resetinterval: sets the time in seconds between restart of unused channels, defaults to 3600 minimum 60 seconds. Some PBXs don't like channel restarts. so set the interval to a very long interval e.g. 1 or 'never' to disable *entirely*. Already did that, this is my zapata.conf :- [channels] switchtype=euroisdn pridialplan=national resetinterval=never usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0 txgain=0 echocancel=yes echocancelwhenbridged=yes ;echocancel=no ;echocancelwhenbridged=no ;echotraining=no ;faxdetect=both ;faxdetect=incoming ;faxdetect=outgoing ;faxdetect=yes faxdetect=no group=1 ;signalling=pri_cpe signalling=pri_net context=incoming channel = 1-15,17-31 jbenable=yes jbforce=yes jbmaxsize=200 ;jbimpl=adaptive ;jbimpl=fixed ;jbresyncthreshold = 4000 group=2 signalling=pri_cpe context=outgoing channel = 32-46,48-62 ;jbenable=yes ;jbforce=yes ;jbmaxsize=200 And my zaptel.conf :- span=1,0,0,ccs,hdb3,crc4 span=2,1,0,ccs,hdb3,crc4 #Span 1 Channel Definition bchan=1-15,17-31 #hardhdlc=16 dchan=16 #Span 2 Channel Definition bchan=32-46,48-62 #hardhdlc=47 dchan=47 loadzone = uk defaultzone = uk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ztd-ethmf
I expected to find th module ztd-ethmf[.c...] in support of the redfone TDMoE product in my zaptel distro (I have 1.4.11). But it's not there. I am awaiting a response to a trouble ticket from redfone. Can anyone give me a jumpstart? I can't seem to google this up. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime pounds MySQL
On Thursday 21 August 2008 10:08:53 J.M. wrote: I am running Asterisk 1.4.21.2 with Realtime. I have a phone setup in the database and when I connect that phone to Asterisk there are suddenly an endless number of SELECT * FROM sip WHERE name = '1001' AND host = 'dynamic' queries being run. The only way to stop the flood of queries coming from Asterisk to restart the Asterisk process. Even disconnecting the phone doesn't stop Asterisk from running the queries. Has anyone seen this before? Why would Asterisk do that and does anyone know the fix? Asterisk does that because realtime data is not cached by default, so for each access of the peer in question, Asterisk needs to reload the data on the peer from the database. If you'd like, turn on rtcachefriends in sip.conf, which will cache the peer for the duration of the registration interval (or whatever you have rtexpire set to). Also, to get correct behavior on reload, you'll need to have rtupdate turned on. Some of the behavior isn't quite right in 1.4.21.2, even, but it should be fixed once 1.4.22 is released. BTW, I would otherwise have responded sooner, but I am on vacation this week, and I am not responding to email as quickly as I would usually. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] set callerid with plus sign
Hi Thanks for all your reply. Just figured out that ISUP does not decode plus sign very well. regards nhadie Eric ManxPower Wieling wrote: + is not a valid Caller*ID character. Asterisk allows you to use + in Caller*ID, but many carriers will reject the call if you do that. Benny Amorsen wrote: ronald [EMAIL PROTECTED] writes: Is it possible to assign a plus sign on the callerid(num) ? Yes. currently this is what i do CALLERID(num)=+6523450017 but telco is denying calls, coz they said they are seeing bs523450017 instead of +6523450017. Which techology? SIP? PRI? POTS? ...? /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with modem data calls and xorcom astribanks
Darren Sessions wrote: Not sure what you've heard before, but I have successfully used a modem at 9600 baud Well, OK, it won't work was a little strong. Faxes work because they too are at slow speed. But for me at least, 9600 baud is pretty much useless. Instead, I just patch the modem directly into the wall plate (bypassing the asterisk box) on the rare occasions that I need to dial up so that I can get the 56k. Works in my small home setup, won't work for everyone. I suspect that neither patching around asterisk nor settling for 9600 is going to be acceptable for the motel manager. --Greg ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] frequent channel reset problem
--- On Fri, 8/22/08, Ming-Ching Tiew [EMAIL PROTECTED] wrote: From: Ming-Ching Tiew [EMAIL PROTECTED] Subject: Re: [asterisk-users] frequent channel reset problem To: asterisk-users@lists.digium.com Date: Friday, August 22, 2008, 11:00 PM Sorry to have posted reply to your email directly. Repost to maillist. resetinterval already set to never to start with. Thanks --- On Fri, 8/22/08, Ming-Ching Tiew [EMAIL PROTECTED] wrote: From: Ming-Ching Tiew [EMAIL PROTECTED] Subject: Re: [asterisk-users] Fw: [asterisk-dev] frequent channel reset problem To: Steve Totaro [EMAIL PROTECTED] Date: Friday, August 22, 2008, 10:58 PM --- On Fri, 8/22/08, Steve Totaro [EMAIL PROTECTED] wrote: From: Steve Totaro [EMAIL PROTECTED] Try setting resetinterval to never. resetinterval: sets the time in seconds between restart of unused channels, defaults to 3600 minimum 60 seconds. Some PBXs don't like channel restarts. so set the interval to a very long interval e.g. 1 or 'never' to disable *entirely*. Already did that, this is my zapata.conf :- [channels] switchtype=euroisdn pridialplan=national resetinterval=never usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0 txgain=0 echocancel=yes echocancelwhenbridged=yes ;echocancel=no ;echocancelwhenbridged=no ;echotraining=no ;faxdetect=both ;faxdetect=incoming ;faxdetect=outgoing ;faxdetect=yes faxdetect=no group=1 ;signalling=pri_cpe signalling=pri_net context=incoming channel = 1-15,17-31 jbenable=yes jbforce=yes jbmaxsize=200 ;jbimpl=adaptive ;jbimpl=fixed ;jbresyncthreshold = 4000 group=2 signalling=pri_cpe context=outgoing channel = 32-46,48-62 ;jbenable=yes ;jbforce=yes ;jbmaxsize=200 And my zaptel.conf :- span=1,0,0,ccs,hdb3,crc4 span=2,1,0,ccs,hdb3,crc4 #Span 1 Channel Definition bchan=1-15,17-31 #hardhdlc=16 dchan=16 #Span 2 Channel Definition bchan=32-46,48-62 #hardhdlc=47 dchan=47 loadzone = uk defaultzone = uk When I re-read my post, there is a danger that people thought that resetinterval fixes my problem. That's not what I meant. What I meant to say was that, the resetinterval was all the time set as 'never' to begin with and the frequent channel reset problem was observed with 'resetinterval' set as 'never'. What I really need is someone who knows how to read the pri debug at http://www.geocities.com/mctiew/asterisk/bd2.txt, and perhaps can tell me what wrong with the said PRI line giving this problem ***ONLY*** to asterisk but not giving any problem to other PBXes while the same asterisk settings I have used was working perfectly on another PRI from the same network provider. Perhaps someone could tell me how to increase the tolerance level of asterisk towards PRI lines with lower quality. Regards. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
