[Asterisk-Users] when * start at bootup chan_h323 fails to load
Hi Gurus I am trying to make asterisk load as a linux servics at boot time. I tried both methods; (a) /etc/init.d/asterisk (b) /etc/rc.d/rc.local But * failed to start. What is interesting is the message log (attached below), in either case problem is with chan_h323.so. Which is failing to load. Once the box is booted up I can start * no problems, I can run same asterisk script I used in (a) above and have no problems. Chan_h323 has no complains. So, what could be the difference at boot-time and when I manually run the same script later on ? Here is the log; Dec 23 23:33:50 WARNING[1074494176]: File cdr_addon_mysql.c, Line 258 (my_load_module): MySQL database sock file not specified. Using default Dec 23 23:33:50 WARNING[1074494176]: File chan_iax2.c, Line 5466 (set_config): Ignoring port for now Dec 23 23:33:51 WARNING[1142106560]: File chan_oss.c, Line 238 (sound_thread): Read error on sound device: Resource temporarily unavailable Dec 23 23:33:51 WARNING[1074494176]: File chan_zap.c, Line 7341 (setup_zap): Ignoring rxwink Dec 23 23:33:52 WARNING[1074494176]: File loader.c, Line 239 (ast_load_resource): libpt_linux_x86_r.so.1.5.2: cannot open shared object file: No such file or directory Dec 23 23:33:52 WARNING[1074494176]: File loader.c, Line 407 (load_modules): Loading module chan_h323.so failed! Any help greatly appreciated !!! SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fw: FAX detection Problem
Hi, I am using asterisk with PRI TE410P card. Everything work fine, except that every time I receive a call, I get File chan_zap.c, Line 3546 (zt_read): Fax detected although they are just normal calls. How can i set the threshold of fax detection. What might be wrong that tone_detect function always detect a fax tone. Help please Hisham. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Quality Survey.... :P
I have about a dozen Budgetone 101s and I'm pretty much satisfied with them. Sorry, Brian; they've got some rough edges, but they're $65, for God's sake. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Quality Survey.... :P
On Wed, 2003-12-24 at 08:18, Brian Capouch wrote: I have about a dozen Budgetone 101s and I'm pretty much satisfied with them. Sorry, Brian; they've got some rough edges, but they're $65, for God's sake. They are $65 yes, but you can get the best analog phones on the market for that price and use an ata. If GS could give the information for people on asterisk to develop iax this $65 phone could be even better than most of the phones in the market more features less buggy and cheaper than all the other sip phones out there Miguel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Quality Survey.... :P
So, you can get a really good analog phone for $65, then you mention and use an ata... what does this ATA cost? $65 for the complete set is what I pay for. At that price, I expect an issue here and there. It is still getting the bugs worked out. I don't have the money to buy $300 Cisco phones. quote who=Miguel Cavazos They are $65 yes, but you can get the best analog phones on the market for that price and use an ata. If GS could give the information for people on asterisk to develop iax this $65 phone could be even better than most of the phones in the market more features less buggy and cheaper than all the other sip phones out there -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] caninvite...
hi guys just got a question, im using grandstream phones with canreinvite=no or woteva, all nat etc is working perfectly. but i believe because of the canreinvite, when a call has taken place the voice will be proxied via the sip server to the 2 parties involved. ( which means the sip server is downloading/uploading to each party constantly). Im just curious though with this setup for all clients.. so everything goes through the sip server, how many phone calls do you rekon asterisk could handle if it was say dual 2g or something like that ? I think i read somewere else it was like 60-90 i forget.. but i think that was if rtp was being handled properly. Thanks heaps guys Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Quality Survey.... :P
Miguel Cavazos wrote: On Wed, 2003-12-24 at 08:18, Brian Capouch wrote: I have about a dozen Budgetone 101s and I'm pretty much satisfied with them. Sorry, Brian; they've got some rough edges, but they're $65, for God's sake. They are $65 yes, but you can get the best analog phones on the market for that price and use an ata. Well you certainly could. And you'd then have to add the cost of the ATA to your cost per seat, at least doubling the $65 figure--tripling it if you meant a Cisco ATA. I'd love it as much as the next person if GS would open up the platform, but that's not likely to happen soon. And for all the griping one sees on the part of a few list members, I don't know how one can escape the fact that given their place in the market--at the very bottom--they're pretty functional for the few dollars one has to part with to get hold of one. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Quality Survey.... :P
It is unfortunate that Cisco is so damned expensive. $300 is only the tip of the iceberg if you go the official route... You still haven't paid for their ongoing maintenance. They should really consider selling their phones at a better price. cameron. On Wed, 24 Dec 2003, Robert Hajime Lanning wrote: So, you can get a really good analog phone for $65, then you mention and use an ata... what does this ATA cost? $65 for the complete set is what I pay for. At that price, I expect an issue here and there. It is still getting the bugs worked out. I don't have the money to buy $300 Cisco phones. quote who=Miguel Cavazos They are $65 yes, but you can get the best analog phones on the market for that price and use an ata. If GS could give the information for people on asterisk to develop iax this $65 phone could be even better than most of the phones in the market more features less buggy and cheaper than all the other sip phones out there ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] caninvite...
vocalvoip wrote: hi guys just got a question, im using grandstream phones with canreinvite=no or woteva, all nat etc is working perfectly. but i believe because of the canreinvite, when a call has taken place the voice will be proxied via the sip server to the 2 parties involved. ( which means the sip server is downloading/uploading to each party constantly). Im just curious though with this setup for all clients.. so everything goes through the sip server, how many phone calls do you rekon asterisk could handle if it was say dual 2g or something like that ? I think i read somewere else it was like 60-90 i forget.. but i think that was if rtp was being handled properly. Thanks heaps guys Justin Yes that is true, All traffic will go via the Asterisk server.. As for how many channels, this would depend on your codec.. you would be able to handle far more channels using G.711 than if you were using GSM or G.729.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk MGCP register
Karl Putland wrote: On Sat, 2003-12-20 at 03:22, Senad Jordanovic wrote: Hi, I am trying to figure out if * can register as a client on a remote MGCP service. Just like SIP and other protocols Do. Anyone tried this? No I don't believe it can. The MGCP implementation in Asterisk is a CallAgent not a UserAgent. --Karl Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Thanks, Karl for your answer. I was suspecting that was the case, but wanted to confirm it. Cheers. SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: FWD Holiday Promotion: Free Calling to 8 Countries
I know this is OT for this list, but I havnt seen it mentioned here and in the spirit of 'open source' I thought this would be interesting for readers here: - Original Message - From: Jeff Pulver [EMAIL PROTECTED] Sent: Tuesday, December 23, 2003 11:28 PM Subject: [FWD] FWD Holiday Promotion: Free Calling to 8 Countries Hi There, In the spirit of the holiday season, from today until the end of the year, it is now possible to use Free World Dialup to place, for free, calls into: Australia, Canada, China, Germany, Israel, Italy, United States and the UK. Note: Mobile calls can only be placed to people in the USA and Canada. To place a call, dial: * [country code] number on Free World Dialup. For example: -- USA/Canada: *1 Australia: *61 China: *86 Germany: *49 Italy: *39 Israel: *972 UK: *44 -- My hope is that our promotion will help some families and friends stay in closer touch during this holiday season. Please feel free to let others know about this. I'd appreciate your help in spreading the word and sharing the holiday spirit. :-) Best regards, Jeff p.s. I'm still working on getting the FWD list formally restored. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] offtopic: possible asterisk meeting saturday amsterdam
hello everyone, theres a bi-monthly computer fair in amsterdam on saturday and it looks like a few asterisk users will be attending, and hopefully some more might be able to turn up. admittedly this probably is a bad idea to advertise because the more asterisk people the less likely i am to find cheap AVM Fritz ISDN cards - but what the hell. if you feel like a chat, and want to avoid the family on saturday. myself, fuzzycat and a few others will be at the RAI from around 10am till 1pm. http://www.pcdumpdag.nl/ email me if you think you might make it - and i'll give you my mobile number so we can try and arrange where to meet. sorry for the completely offtopic message. duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MSN to GS - Call drops in 10 secs
i tried with other softphones. the only phone thats working with GS is Xtern. MSN and SJ doesnt work. Is this a known issue. Thanks, -B - Original Message - From: Balaji NJL To: [EMAIL PROTECTED] Sent: Tuesday, December 23, 2003 7:05 PM Subject: Re: [Asterisk-Users] MSN to GS - Call drops in 10 secs i tried with only GSM too. With only GSM it doesnt even connect to GS. Then someone recommended to use ulaw and alaw and that helped. But the call drops after 10 secs. i did a 'sip debug' and what i found is that MSN doesnt even recognize that call is in progress and then drops the call. Any way i can increase this or disable this option. thanks, -B - Original Message - From: Craig Waddington To: [EMAIL PROTECTED] Sent: Tuesday, December 23, 2003 4:34 PM Subject: RE: [Asterisk-Users] MSN to GS - Call drops in 10 secs Balaji, I also have the same issue. Works fine on any phone except GS for me. After a bit of research I found a post saying set the phone to offer only one codec set. It looks like we have to set the phone to use one codec GSM I am concerned that you cant use passwords when logging in to * using Messenger. Craig. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Balaji NJLSent: 23 December 2003 23:04To: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] MSN to GS - Call drops in 10 secs resending. Can anyone help me in trying to understand what would be the problem. appreciate ur time. i need to get this working. thanks a lot, -B - Original Message - From: Balaji NJL To: [EMAIL PROTECTED] Sent: Monday, December 22, 2003 8:15 PM Subject: [Asterisk-Users] MSN to GS - Call drops in 10 secs Hi All, i dont know what changes i made recently but i am unable to hold the call for more 10 secs between MSN and GS. PSTN to MSN and PSTN to GS and vice versa works fine. i am not behind NAT.Also MSN to MSN works fine too. my SIP details [general]port = 5060bindaddr = 0.0.0.0context = bogon-calls;context = defaultdisallow=allallow=ulawallow=alawallow=ilbcallow=gsm ;My SIP phone - GS[2000]type=friendusername=2000secret=qweqwehost=dynamiccontext=from-sipmailbox=2000dtmfmode=inband ;MSN Msgr[2002]type=friendhost=dynamicinsecure=yesdtmfmode=inband;dtmfmode=rfc2833context=from-sipmailbox=2002;auth=plaintext i did a SIP trace it says Format=UKN CSeq=BYE thanks for the help, -Balaji Do you Yahoo!?Yahoo! Photos - Get your photo on the big screen in Times Square Do you Yahoo!?Yahoo! Photos - Get your photo on the big screen in Times Square Do you Yahoo!?Yahoo! Photos - Get your photo on the big screen in Times Square Do you Yahoo!? Yahoo! Photos - Get your photo on the big screen in Times Square
[Asterisk-Users] Grandstream Quality Survey.... :P
From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Grandstream Quality Survey :P Reply-To: [EMAIL PROTECTED] ... I have 2 of these phones and they work fine for my application. Granted its not the most intensive use and definatly not the most critical users but... With all of the companies that are running into cash problems in the next year I think that the demands for systems that do everything including make coffee will decrease. Basic functionality will take the place of complicated functionality. Granted GS needs to be more responsive but if they are going to maintain a low price level we need to be a bit understanding about the responses If GS phones don't meet your needs then by all means spend more money on some of the other brands. For some of us, GS does meet the requirements and we will continue to use them. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Using asterisk as voicemail with SER
On Wed, 17 Dec 2003, Victor Medrano wrote: i did with cisco callmanager with smdi integration . and h323 . works very well . You got SMDI working with CCM? How? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream Quality Survey.... :P
For the price, the Grandstream is unbeatable value for money. Get firmware version 1.04.26 and you should be fine. This firmware fixes issues our customers had with phone lockups, nat problems, one-way audio, stun problems. Best Wishes Tan www.telappliant.com www.voiptalk.org -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of rnc Info Lists Sent: 24 December 2003 12:09 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Grandstream Quality Survey :P From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Grandstream Quality Survey :P Reply-To: [EMAIL PROTECTED] ... I have 2 of these phones and they work fine for my application. Granted its not the most intensive use and definatly not the most critical users but... With all of the companies that are running into cash problems in the next year I think that the demands for systems that do everything including make coffee will decrease. Basic functionality will take the place of complicated functionality. Granted GS needs to be more responsive but if they are going to maintain a low price level we need to be a bit understanding about the responses If GS phones don't meet your needs then by all means spend more money on some of the other brands. For some of us, GS does meet the requirements and we will continue to use them. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FWD problems
I've been having issues getting FWD to work. I posted this same Q to the
FWD forum (no responses yet), but I was hoping someone here had some insight:
http://yabb.pulver.com/cgi-bin/yabb/YaBB.cgi?board=news;action=display;num=1072263468;start=0#0
I just signed up for an FWD account (I know I had one before, but I lost
it.. :)
I've got it running through Asterisk - all working fine from a SIP
standpoint. I can dial FWD numbers like 612/613/etc and everything
works. However, if I dial *18005551212 or *408xxx (say, a USA number),
I either get a fast busy or a This service is only available to FreeWorld
Dialup members.
Am I missing something? I signed up, got my password .. the sip is
registered, firewall is open, no NAT, etc. I've tried a variety of combos
in dialing/etc .. to no avail. Is my account pending some type of
activation or such?
Possibly / likely related, it seems that the * doesn't work when I'm
trying to set up the voicemail either. I'm using a Cisco 7960 (but
remember, it's actually Asterisk linking us together). The 7960 does have a
* dialplan, so that shouldnt be an issue.
Any ideas you guys have would be great!
Here's what my sip.conf looks like:
register=9:[EMAIL PROTECTED]/453
It shows that it's registered in sip sho reg..
[fwd]
type=friend
secret=password
username=9
fromuser=9 ; I dont need this .. but worth a shot, tried with and
without
nat=yes ;I'm not behind nat, but I thought I'd try it
anyway
fromdomain=fwd.pulver.com ; Don't need this either. .but what the hay
host=fwd.pulver.com
canreinvite=no ; worth a shot, right?
reinvite=no
I then have an extension that does:
exten = _7.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
-d
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RE: [Asterisk-Users] Grandstream Quality Survey.... :P
Hi Tan, Can you supply us with 1.0.4.26 firmware? Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: 24 December 2003 12:53 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Grandstream Quality Survey :P For the price, the Grandstream is unbeatable value for money. Get firmware version 1.04.26 and you should be fine. This firmware fixes issues our customers had with phone lockups, nat problems, one-way audio, stun problems. Best Wishes Tan www.telappliant.com www.voiptalk.org -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of rnc Info Lists Sent: 24 December 2003 12:09 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Grandstream Quality Survey :P From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Grandstream Quality Survey :P Reply-To: [EMAIL PROTECTED] ... I have 2 of these phones and they work fine for my application. Granted its not the most intensive use and definatly not the most critical users but... With all of the companies that are running into cash problems in the next year I think that the demands for systems that do everything including make coffee will decrease. Basic functionality will take the place of complicated functionality. Granted GS needs to be more responsive but if they are going to maintain a low price level we need to be a bit understanding about the responses If GS phones don't meet your needs then by all means spend more money on some of the other brands. For some of us, GS does meet the requirements and we will continue to use them. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream Quality Survey.... :P
On Wed, 2003-12-24 at 14:35, David J Carter wrote: Hi Tan, Can you supply us with 1.0.4.26 firmware? http://www.grandstream.com/TEMP/FIRMWARE/ -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream Quality Survey.... :P
Try it on one of the phones first. We've tested it and it seems to work fine. Let me know offline how you get on. http://www.telappliant.com/grandstream/1.04.26.zip Thanks Tan www.telappliant.com www.voiptalk.org -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David J Carter Sent: 24 December 2003 13:36 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Grandstream Quality Survey :P Hi Tan, Can you supply us with 1.0.4.26 firmware? Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: 24 December 2003 12:53 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Grandstream Quality Survey :P For the price, the Grandstream is unbeatable value for money. Get firmware version 1.04.26 and you should be fine. This firmware fixes issues our customers had with phone lockups, nat problems, one-way audio, stun problems. Best Wishes Tan www.telappliant.com www.voiptalk.org -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of rnc Info Lists Sent: 24 December 2003 12:09 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Grandstream Quality Survey :P From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Grandstream Quality Survey :P Reply-To: [EMAIL PROTECTED] ... I have 2 of these phones and they work fine for my application. Granted its not the most intensive use and definatly not the most critical users but... With all of the companies that are running into cash problems in the next year I think that the demands for systems that do everything including make coffee will decrease. Basic functionality will take the place of complicated functionality. Granted GS needs to be more responsive but if they are going to maintain a low price level we need to be a bit understanding about the responses If GS phones don't meet your needs then by all means spend more money on some of the other brands. For some of us, GS does meet the requirements and we will continue to use them. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream Quality Survey.... :P
http://www.grandstream.com/TEMP/FIRMWARE/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David J Carter Sent: Wednesday, December 24, 2003 7:36 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Grandstream Quality Survey :P Hi Tan, Can you supply us with 1.0.4.26 firmware? Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: 24 December 2003 12:53 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Grandstream Quality Survey :P For the price, the Grandstream is unbeatable value for money. Get firmware version 1.04.26 and you should be fine. This firmware fixes issues our customers had with phone lockups, nat problems, one-way audio, stun problems. Best Wishes Tan www.telappliant.com www.voiptalk.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FWD problems
On Wed, 24 Dec 2003, denon wrote:
I've been having issues getting FWD to work. I posted this same Q to the
FWD forum (no responses yet), but I was hoping someone here had some insight:
My setup is like this:
sip.conf:
register = 21542:[EMAIL PROTECTED]/6002 ; Free World Dialup
[fwd.pulver.com]
type=peer
host=fwd.pulver.com
fromuser=21542
fromdomain=fwd.pulver.com
username=21542
secret=password
In extensions.conf:
; Free World Dialup
[fwd]
exten = _10113.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _10113.,2,Congestion
(I use a 10113 prefix for FWD numbers).
We're chatting to friends in the UK right now so seems to work for me.
Steve
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RE: [Asterisk-Users] Merry Christmas, all Asterisk users!
merry christmas to all. sorry (but come the 1st it's a new year and therefore can create a new list to atone for) G Happy Holidays Everyone! May your uptimes be plentiful, and your core dumps be rare in this season of hardware failure. And may those who still use the old school 'G' notation never vanish from this earth. -- Tony Kava Network Administrator Pottawattamie County, Iowa ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Quality Survey.... :P
I just loaded the b13p4.30.zip firmware and now I am not able to log into the GS admin interface.. anyone else having this problem? Going to try the next older version.. Later.. mikeu wrote: http://www.grandstream.com/TEMP/FIRMWARE/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David J Carter Sent: Wednesday, December 24, 2003 7:36 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Grandstream Quality Survey :P Hi Tan, Can you supply us with 1.0.4.26 firmware? Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: 24 December 2003 12:53 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Grandstream Quality Survey :P For the price, the Grandstream is unbeatable value for money. Get firmware version 1.04.26 and you should be fine. This firmware fixes issues our customers had with phone lockups, nat problems, one-way audio, stun problems. Best Wishes Tan www.telappliant.com www.voiptalk.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Quality Survey.... :P
I have 6 broken Grandstreams- out of an order of 8. After having tested over a dozen IP phone products, I found that Grandstream was the worst choice of the group. I would never recommend that anyone buys this product unless they are using it for non-essential use. To put it simply: Grandstream phones are complete crap. -GSR - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 24, 2003 6:43 AM Subject: [Asterisk-Users] Grandstream Quality Survey :P Today class we are going to be talking about the wonderful line of GrandStream products. Or should I say BarbieTone phones? Who else is having MAJOR issues with the grandstream products? How many times have you been told upgrade upgrade upgrade? How many of you have paperweights, granted the phone is light as a feather and couldn't weight papers down in the first place? How about that ring tone, really dandy eh? Who else is irked about the the GAPS crap? It should slurp down cfgMACADDRESS.txt and we shouldn't have to pay more for that option. Have you had Message Waiting Indicator issues? Have you had issues with the Hold button and flash button? Have you had issues with sip transfers? Has the grandstream product line made you want to hurt someone? Care for some matches and lighter fluid? Was the response from grandstream support able to take care of your problems? I own a grandstream phone and I guess I just don't use it enought to see alot of these problems but the consensus on #asterisk is they are CRAP and everyone should stop buying them till they get their act together. A few people in the asterisk community have offered to write IAX firmware for the phones but grandstream has give them the run around. If they can't create stable and usable firmware they should atleast let the info out to let someone write IAX firmware for the damn thing. BOYCOTT GRANDSTREAM Thanks, bkw_ PS: then again you get what you pay for, 10 dollar phone with a 65 dollar pricetag. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Merry Christmas, all Asterisk users!
Thanks! And may your CDR's be longer and more profitable!!! :) Merry Christmas everyone! Panny Malialis Hotlinks Internet Services http://www.hotlinks.co.uk - Original Message - From: Tony Kava [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 24, 2003 2:30 PM Subject: RE: [Asterisk-Users] Merry Christmas, all Asterisk users! merry christmas to all. sorry (but come the 1st it's a new year and therefore can create a new list to atone for) G Happy Holidays Everyone! May your uptimes be plentiful, and your core dumps be rare in this season of hardware failure. And may those who still use the old school 'G' notation never vanish from this earth. -- Tony Kava Network Administrator Pottawattamie County, Iowa ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Quality Survey.... :P
On Wed, 2003-12-24 at 15:50, WipeOut wrote: I just loaded the b13p4.30.zip firmware and now I am not able to log into the GS admin interface.. anyone else having this problem? Yep been there. Panicked, rebooted the phone and it responded as normal. I just tried it again because of your question and had to reboot again to get in. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] when * start at bootup chan_h323 fails to load
SW wrote: (ast_load_resource): libpt_linux_x86_r.so.1.5.2: cannot open shared object file: No such file or directory Dec 23 23:33:52 WARNING[1074494176]: File loader.c, Line 407 (load_modules): Loading module chan_h323.so failed! RTFM cat /path/to/asterisk/channels/h323/README Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Quality Survey.... :P
Well you certainly could. And you'd then have to add the cost of the ATA to your cost per seat, at least doubling the $65 figure--tripling it if you meant a Cisco ATA. NOT, Cisco ATA's can be had fro 99-120 if you are lucky. Then you can also get a cisco 7905 which can be had on ebay for about 99 bucks or so. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Quality Survey.... :P
Cisco 7905's are damn fine phones for 99 bucks and they blow the grandstream away... bkw On Wed, 24 Dec 2003, Cameron Palmer wrote: It is unfortunate that Cisco is so damned expensive. $300 is only the tip of the iceberg if you go the official route... You still haven't paid for their ongoing maintenance. They should really consider selling their phones at a better price. cameron. On Wed, 24 Dec 2003, Robert Hajime Lanning wrote: So, you can get a really good analog phone for $65, then you mention and use an ata... what does this ATA cost? $65 for the complete set is what I pay for. At that price, I expect an issue here and there. It is still getting the bugs worked out. I don't have the money to buy $300 Cisco phones. quote who=Miguel Cavazos They are $65 yes, but you can get the best analog phones on the market for that price and use an ata. If GS could give the information for people on asterisk to develop iax this $65 phone could be even better than most of the phones in the market more features less buggy and cheaper than all the other sip phones out there ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream Quality Survey.... :P
Unbeatable maybe... but also very unreliable. On Wed, 24 Dec 2003 [EMAIL PROTECTED] wrote: For the price, the Grandstream is unbeatable value for money. Get firmware version 1.04.26 and you should be fine. This firmware fixes issues our customers had with phone lockups, nat problems, one-way audio, stun problems. Best Wishes Tan www.telappliant.com www.voiptalk.org -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of rnc Info Lists Sent: 24 December 2003 12:09 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Grandstream Quality Survey :P From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Grandstream Quality Survey :P Reply-To: [EMAIL PROTECTED] ... I have 2 of these phones and they work fine for my application. Granted its not the most intensive use and definatly not the most critical users but... With all of the companies that are running into cash problems in the next year I think that the demands for systems that do everything including make coffee will decrease. Basic functionality will take the place of complicated functionality. Granted GS needs to be more responsive but if they are going to maintain a low price level we need to be a bit understanding about the responses If GS phones don't meet your needs then by all means spend more money on some of the other brands. For some of us, GS does meet the requirements and we will continue to use them. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Quality Survey.... :P
Yep I have heard this many many times. Seems like they have a large batch of phones that are bad. bkw On Wed, 24 Dec 2003, Greg Renouf wrote: I have 6 broken Grandstreams- out of an order of 8. After having tested over a dozen IP phone products, I found that Grandstream was the worst choice of the group. I would never recommend that anyone buys this product unless they are using it for non-essential use. To put it simply: Grandstream phones are complete crap. -GSR - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 24, 2003 6:43 AM Subject: [Asterisk-Users] Grandstream Quality Survey :P Today class we are going to be talking about the wonderful line of GrandStream products. Or should I say BarbieTone phones? Who else is having MAJOR issues with the grandstream products? How many times have you been told upgrade upgrade upgrade? How many of you have paperweights, granted the phone is light as a feather and couldn't weight papers down in the first place? How about that ring tone, really dandy eh? Who else is irked about the the GAPS crap? It should slurp down cfgMACADDRESS.txt and we shouldn't have to pay more for that option. Have you had Message Waiting Indicator issues? Have you had issues with the Hold button and flash button? Have you had issues with sip transfers? Has the grandstream product line made you want to hurt someone? Care for some matches and lighter fluid? Was the response from grandstream support able to take care of your problems? I own a grandstream phone and I guess I just don't use it enought to see alot of these problems but the consensus on #asterisk is they are CRAP and everyone should stop buying them till they get their act together. A few people in the asterisk community have offered to write IAX firmware for the phones but grandstream has give them the run around. If they can't create stable and usable firmware they should atleast let the info out to let someone write IAX firmware for the damn thing. BOYCOTT GRANDSTREAM Thanks, bkw_ PS: then again you get what you pay for, 10 dollar phone with a 65 dollar pricetag. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Quality Survey.... :P
Yes when you upgrade to beta code you may have to reboot 3 times for the phone to function properly. Then cross your fingers that the phone will accually register with * once you do that. bkw On Wed, 24 Dec 2003, Dave Cotton wrote: On Wed, 2003-12-24 at 15:50, WipeOut wrote: I just loaded the b13p4.30.zip firmware and now I am not able to log into the GS admin interface.. anyone else having this problem? Yep been there. Panicked, rebooted the phone and it responded as normal. I just tried it again because of your question and had to reboot again to get in. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FWD problems
--On Wednesday, December 24, 2003 07:12:05 -0600 denon [EMAIL PROTECTED] wrote: I've got it running through Asterisk - all working fine from a SIP standpoint. I can dial FWD numbers like 612/613/etc and everything works. However, if I dial *18005551212 or *408xxx (say, a USA number), I either get a fast busy or a This service is only available to FreeWorld Dialup members. I have exactly this problem and posted a bug report to FWD about a week ago - no response yet. It's bizarre that FWD recognises you to dial another user but not to call outside their network. Sounds more like a FWD problem than a * problem to me. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Quality Survey.... :P
Brian West wrote: Well you certainly could. And you'd then have to add the cost of the ATA to your cost per seat, at least doubling the $65 figure--tripling it if you meant a Cisco ATA. NOT, Cisco ATA's can be had fro 99-120 if you are lucky. Then you can also get a cisco 7905 which can be had on ebay for about 99 bucks or so. bkw Brian, The price you can get a phone for on ebay is not relevant to a business only to a home user.. If I have to quote a customer for a phone I can't tell him that the price is $100 - $120 and the availibility is unknown.. Whet if I have to get 20 or more of them?? How long would it take me to buy a number of them off ebay?? Add to that the fact thet there are not that many availible on the UK ebay.. At the end of the day IMO you have to use the new purchase price of the phone when making product comparisons.. With that as a factor the Cisco looses lots of points.. I agree that the GS is not proving to be all that good in terms of reliability and ease of use in the real world but it can't be touched on cost.. I don't see Cisco dropping their prices so they will always be and enterprise only product.. Maybe Snom will drop their prices and become more competitive in the budget arena where the GS currently holds the crown.. I think the Snom phones are great and if the price were lower thay would have a good chance at market domination in the SOHO and small to medium business space.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Weirdness with CALLERID / CALLERIDNAME from incoming PRI
Hey all,
We've upgraded our PRI trunk to support what BellSouth calls enhanced
caller id name delivery. The weird part is, I'm only capable of seeing
the name portion on incoming calls within voicemail2's email delivery.
For example, on an incoming call, asterisk is reporting this:
Context from extensions.conf (BS delivers 7-digit DIDs):
exten = 9133727,1,Answer
exten = 9133727,2,SetMusicOnHold,random
exten = 9133727,3,NoOp,${CALLERID}
exten = 9133727,4,Dial(SIP/5001SIP/5013,20)
exten = 9133727,5,Voicemail2([EMAIL PROTECTED])
exten = 9133727,105,Dial(SIP/5002)
exten = 9133727,106,Voicemail2([EMAIL PROTECTED])
exten = 9133727,206,Voicemail2([EMAIL PROTECTED])
When calling from 404-555-8183 to the number above the console is
reporting:
-- Accepting call from '4045558183' to '9133727' on channel 6, span
1
-- Executing SetMusicOnHold(Zap/6-1, random) in new stack
-- Executing NoOp(Zap/6-1, 4049338183) in new stack
So from the NoOp echo, I don't see the full name (fyi, I get the same
for CALLERIDNAME also).
However, if I let the call go through to voicemail via priority 5, the
email message I receive has the following header:
Just wanted to let you know you were just left a 0:05 long
message (number 1) in mailbox 5001 from ATLANTA, GA 4045558183, on
Wednesday, December 24, 2003 at 10:18:22 AM so you might want to check
it when you get a chance. Thanks!
My thought is that the name is being delivered else voicemail2 wouldn't
be able to get the ATLANTA, GA portion.
Running from CVS, version: Asterisk CVS-12/24/03-10:55:24
Any idea why I'm not able to parse or see the CALLERIDNAME from
extensions.conf but voicemail2 sees it just fine?
Happy Holidays all,
--- Gavin Adams
Promisant (Technology) Ltd.
Atlanta, GA
___
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RE: [Asterisk-Users] Grandstream Quality Survey.... :P
Brian, Can you compare Ford and Mercedes or BMW? Both are cars and drives.. but you have different feeling and price in/for each car ..same here Grandstream is low-cost solution for end-users/small business , Cisco IP Phones are couple times more expensive ,but they have more features, less bugs and more fancy. Also, don't forget that Grandstream is muuuch smaller company compare to Cisco and they are new company, they have much less customers/phones sold out then Cisco, so it takes time to find all bugs and fix them, also to release new firmware. We had conversation with Grandstream how to improve there phones, so they are working on it. I am sure in 2004 GS will go high and we will have less probs with them. We are looking now to improve GS products and start collecting all bugs/probs and send them to GS. Idea is that we are opening Online forums and special Grandstream products mailing list. Some support people from Grandstream will be participating in Forums and Mailing lists, so we will have direct communication between GS and Online community, hopefully it will help us to solve more probs. Grandstream is very interested to make nice product and sell more, so they will be fixing bugs for sure, otherwise they will be out of business. Grandstream forums URL : http://forum.xvoip.com/viewforum.php?f=7 Grandstream products support mailing list: [EMAIL PROTECTED] Regards, Alexander -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Wednesday, December 24, 2003 10:32 AM To: Asterisk List Subject: Re: [Asterisk-Users] Grandstream Quality Survey :P Yes when you upgrade to beta code you may have to reboot 3 times for the phone to function properly. Then cross your fingers that the phone will accually register with * once you do that. bkw On Wed, 24 Dec 2003, Dave Cotton wrote: On Wed, 2003-12-24 at 15:50, WipeOut wrote: I just loaded the b13p4.30.zip firmware and now I am not able to log into the GS admin interface.. anyone else having this problem? Yep been there. Panicked, rebooted the phone and it responded as normal. I just tried it again because of your question and had to reboot again to get in. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] amaflags question
I am trying to configure cdr on a system. We are using nufone and I have set amaflags=billing on both of their sections in iax.conf. Incoming nufone calls show up in cdr with billing, but outgoing calls still show documentation. What do I need to change? We have a handful of SIP phones, 1 X100P outside line for local, and the rest is via nufone. I don't want inter-system calls to be marked for billing and I don't want local calls to be marked for billing. Thanks dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CT1 and callerid / DNIS
On Tue, 2003-12-23 at 19:22, Brian West wrote: I'm just double checking.. I was told it wasn't possible but i'm going to ask just in case. Can you set outbound callerid on a channelized T1? I think there is a way to do something like DID with the 4 digits ofDTMF passed before the call. It is unlikely though that you will findsomeone interested in doing that though. It is easier/cheaper to drop aPRI into somewhere and then outbound caller ID isn't kludgey with DTMF. -- Steven Critchfield [EMAIL PROTECTED] The service you might be referring to is Dialed Number Identification Service (DNIS) that is put on T1's for inbound 800 and 900 lines. This is an inband delivery of the last 4-digits of a dialed number (800/900) that is passed into the PBX from the SPfor callcenter or other routing. Does Asterisk support this? - David Schlossman ([EMAIL PROTECTED])
RE: [Asterisk-Users] Grandstream Quality Survey.... :P
Message: 11 From: Asterisk online forums [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Grandstream Quality Survey :P Date: Wed, 24 Dec 2003 11:23:14 -0500 Reply-To: [EMAIL PROTECTED] Brian, ... We are looking now to improve GS products and start collecting all bugs/probs and send them to GS. Idea is that we are opening Online forums and special Grandstream products mailing list. Some support people from Grandstream will be participating in Forums and Mailing lists, so we will have direct communication between GS and Online community, hopefully it will help us to solve more probs. Grandstream is very interested to make nice product and sell more, so they will be fixing bugs for sure, otherwise they will be out of business. Alexander, I agree with your email but setting up MORE forums and mailing lists is not productive. GS phones have problems interacting with the VoIP services and Asterisk. The BEST places for the GS folks to get feedback AND to interact with the people who are using their phones are on these already existing mailing lists. I don't know why you insist on creating even more websites/email lists for VoIP support. Why not encourage GS to get visible on these lists and interact with their customers here, where they can get the most concentrated feedback (good and bad). Also, a comment for the general list. To me BETA code means that it is NOT yet RELEASED as PRODUCTION code. For anyone to think that Beta code comes without problems is being a bit shortsighted. If you get beta code that works without problems then that is great, otherwise give the developer feedback so that he can fix the bugs and don't complain about the problems it caused you. Otherwise wait on the official production releases. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream budgetTone registration time out
hi, i have been using grandstream budgettone IP phones and they work fine except that these phones times out after some hours.. i ahve seen that the phones working ok are next day unregistered and sip show peers do not show their IP and although these phones can make calls , they cannot be called. They Sip show peers only shows their IP when i restart the IP phones. This is really annoying me now. Is there any better solutions than just restarting the phones every day? Any help is appreciated. cm
[Asterisk-Users] Grandstream 102 flashing display
The phone powers up and I can make calls through my Asterisk gateway to other endpoints. However the four leds under the keypad are permanently illuminated and the backlight slowly flashes on and off. When I pick up the handset there is a repeated tone before I get a dial tone. I know it's trying to tell me something, but the manual does not give anything away. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream 102 flashing display
voicemail notification? -Original Message-From: bam [mailto:[EMAIL PROTECTED]Sent: Wednesday, December 24, 2003 12:17 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Grandstream 102 flashing displayThe phone powers up and I can make calls through my Asterisk gateway to other endpoints. However the four leds under the keypad are permanently illuminated and the backlight slowly flashes on and off. When I pick up the handset there is a repeated tone before I get a dial tone. I know it's trying to tell me something, but the manual does not give anything away. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream budgetTone registration time out
What version of the BudgeTone software are you running? - Original Message - From: Chandra To: [EMAIL PROTECTED] Sent: Wednesday, December 24, 2003 12:09 PM Subject: [Asterisk-Users] Grandstream budgetTone registration time out hi, i have been using grandstream budgettone IP phones and they work fine except that these phones times out after some hours.. i ahve seen that the phones working ok are next day unregistered and sip show peers do not show their IP and although these phones can make calls , they cannot be called. They Sip show peers only shows their IP when i restart the IP phones. This is really annoying me now. Is there any better solutions than just restarting the phones every day? Any help is appreciated. cm
Re: [Asterisk-Users] Grandstream 102 flashing display
The phone powers up and I can make calls through my Asterisk gateway to other endpoints. However the four leds under the keypad are permanently illuminated and the backlight slowly flashes on and off. When I pick up the handset there is a repeated tone before I get a dial tone. I know it's trying to tell me something, but the manual does not give anything away. Can't say for the LEDS being illuminated but a flashing backlight and stutter dialtone is the normal message waiting indicator that the phone gives when Asterisk tells it that a mesasge is waiting... I don't remember the exact syntax in sip.conf since am away from my Asterisk box. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Short in my X100P. Is it broke?
At my home office I have a X100P card in a server that I've been using for testing. The machine it is in is connected to a HP fax machine and then to the wall outlet. This morning the SBC installer showed up at my house for the ADSL install on that line. He said they detected a short. So he tested the outside box and it was fine. He said it was inside. So we came inside and tested the two devices with his little box. The fax was fine... the X100P card however was causing the short. Now of course I'm going to install a filter on this line for the ADSL, but is this short normal? The installer says that it will kill the ADSL signal. Maybe the X100P is defective? It's been working fine though for making and answering calls to this point. Thanks, Jonathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream 102 flashing display
Title: Leterhead Mine does that as a message indicator when mail is in the mailbox. You get a flashing display and a stuttered dial tone for the first few seconds. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of bam Sent: 24 December 2003 17:17 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Grandstream 102 flashing display The phone powers up and I can make calls through my Asterisk gateway to other endpoints. However the four leds under the keypad are permanently illuminated and the backlight slowly flashes on and off. When I pick up the handset there is a repeated tone before I get a dial tone. I know it's trying to tell me something, but the manual does not give anything away. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Registered Office: - 23 First Street, Low Moor, Bradford, West Yorkshire, BD12 0JQ. Company Registration Number: - 03807643. VAT Registration Number: - 734-3363-42 Telephone / Fax: - 44 (0) 7092 154039. SIP_Phone: - 1 (747)669 1957 http://www.codepipe.ltd.uk / http://www.codepipe.com / E-Mail: - [EMAIL PROTECTED]
Re: [Asterisk-Users] Grandstream 102 flashing display
bam wrote: The phone powers up and I can make calls through my Asterisk gateway to other endpoints. However the four leds under the keypad are permanently illuminated and the backlight slowly flashes on and off. When I pick up the handset there is a repeated tone before I get a dial tone. I know it's trying to tell me something, but the manual does not give anything away. You have a new voicemail message!!.. The flashing is the MWI and the dialtoane is called a stutter dial tone which is an alternate way of telling you that there is voicemail.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FWD problems
--On Wednesday, December 24, 2003 07:12:05 -0600 denon [EMAIL PROTECTED] wrote: I've got it running through Asterisk - all working fine from a SIP standpoint. I can dial FWD numbers like 612/613/etc and everything works. However, if I dial *18005551212 or *408xxx (say, a USA number), I either get a fast busy or a This service is only available to FreeWorld Dialup members. I have exactly this problem and posted a bug report to FWD about a week ago - no response yet. It's bizarre that FWD recognises you to dial another user but not to call outside their network. Sounds more like a FWD problem than a * problem to me. Read the fwd announcement. Jeff Pulver mentioned the fact that * users cannot use the free holiday calls, since FWD cannot be sure that * is not being used by more than 1 user at the same time. Arnold ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Quality Survey.... :P
When moving from 1.0.3.x firmware to 1.0.4.x you must reboot 2 times : first time for loading the new bootloader from tftp second time for getting the 1.0.4.x firmware. GS are ok for their price. but of course, you get what you paid for. with 1.0.4.26 firmware I'm quite happy, finally there's early media (you can listen moh while in ring state) and they seems more stable. Of course some features are missing, like supervised transfers , that is *essential* in a business environment, and 10/100 eth port... lowering from 100 to 10 is bad... just my 2 cents. matteo Il mer, 2003-12-24 alle 15:50, WipeOut ha scritto: I just loaded the b13p4.30.zip firmware and now I am not able to log into the GS admin interface.. anyone else having this problem? Going to try the next older version.. Later.. mikeu wrote: http://www.grandstream.com/TEMP/FIRMWARE/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David J Carter Sent: Wednesday, December 24, 2003 7:36 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Grandstream Quality Survey :P Hi Tan, Can you supply us with 1.0.4.26 firmware? Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: 24 December 2003 12:53 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Grandstream Quality Survey :P For the price, the Grandstream is unbeatable value for money. Get firmware version 1.04.26 and you should be fine. This firmware fixes issues our customers had with phone lockups, nat problems, one-way audio, stun problems. Best Wishes Tan www.telappliant.com www.voiptalk.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream Quality Survey.... :P
Robert, We are going to deploy GS phones in our free voice network, therefore we require somekind of web-presence, which will reflect GS support,etc. Unfortunately not all of our users are subscribed to Asterisk mailing list. Acting as GS distributors, we are making separate forum for this, which doesn't belong to Asterisk. My postage about new forum was just as information only about new resource. And I assume asterisk mailing list if primarily designed to Asterisk support and not Grandstream phones... Also Grandstream phones are being used in different platforms too, not necessarily only with Asterisk, this is why they are looking for separate resource. In all cases, I will be posting here copy's of interesting messages/infos from Grandstream, so we all know what's going on. And I agree with your comment on BETA firmware. I assume people have to understand what is Beta release and what is stable official release of software. Of course by using Beta software, which is not approved officially and launched bugs will appear. Regards, Alexander rnc Info Lists Sent: Wednesday, December 24, 2003 12:05 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Grandstream Quality Survey :P ... Alexander, I agree with your email but setting up MORE forums and mailing lists is not productive. GS phones have problems interacting with the VoIP services and Asterisk. The BEST places for the GS folks to get feedback AND to interact with the people who are using their phones are on these already existing mailing lists. I don't know why you insist on creating even more websites/email lists for VoIP support. Why not encourage GS to get visible on these lists and interact with their customers here, where they can get the most concentrated feedback (good and bad). Also, a comment for the general list. To me BETA code means that it is NOT yet RELEASED as PRODUCTION code. For anyone to think that Beta code comes without problems is being a bit shortsighted. If you get beta code that works without problems then that is great, otherwise give the developer feedback so that he can fix the bugs and don't complain about the problems it caused you. Otherwise wait on the official production releases. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream budgetTone registration time out
--- Chandra [EMAIL PROTECTED] wrote: i have been using grandstream budgettone IP phones and they work fine except that these phones times out after some hours.. i ahve seen that the phones working ok are next day unregistered and sip show peers do not show their IP and although these phones can make calls , they cannot be called. They Sip show peers only shows their IP when i restart the IP phones. This is really annoying me now. Is there any better solutions than just restarting the phones every day? Hi. I just got 2 BT101s yesterday. I had a problem with one of mine not being able to call out or receieve calls. I believe that the registration icon was not on the phone. I can't remember what was shown on Asterisk. This issue has come up before. I am behind a Linksys router using NAT/DHCP. I added qualify=60 and have not seen the problem again although I have not had them up very long and not tested them very much. Kevin _ Are you a Techie? Get Your Free Tech Email Address Now! Visit http://www.TechEmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax capabilities of various services
Title: Fax capabilities of various services For the Vonage, Packet8, etc services, are they all able to handle fax machines on their little interconnect boxes? Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses & dangerous content at One Unified and is believed to be clean.
[Asterisk-Users] Sip phones on the same extension?
Hello. I'm a new Asterisk user, but I'm impressed with the flexibility and versatility of Asterisk, and am moving quickly to adopt it's main-line use in our company. Hopefully, you'll be hearing more from me as the project moves forward. Right now, though, I have a question about SIP peer registration. Right now, for our SIP-based phone,s, we're using the Sip Express Router product, which accepts sip registration requests and lets us route calls to any of the phones which register with SER. I am a semi-nomatic user, and can work at any of three different locations. Right now, my phones all sign up with SER, and register with the same telephone number. When someone dials that number, all three phones ring, and which ever one gets answered first, gets the call. When I tried to do this with Asterisk, sources from the cvs repository as of 12/18/2003, sip show peers only showed the most recent registration. This lead me to believe that if I dialed the number, only the most recently registered phone would ring. I was able to work around the problem by defining an umbrella extension which rings all three phones at the same time, but I'd like to have a way of dynamically adding phones to a given extension without having to necessarily rewrite the extensions.conf file, and I'd like calls from these extensions to show up from the master extension that folks should use to reach me. I imagine I could do something with pickup groups, but my understanding is that it is not true that all phones in a pickup group will necessarily ring just because they're a member of a given pickup group. The phones on this particular extension are many miles from each other, so one couldn't hear the other phone ring. Another work around is to put Asterisk behind SER, but this seems overly complicated, and I want to make sure that Asterisk doesn't do what I want before I pursue that path. Any suggestions on how to have multiple phones register with the same number in Asterisk? -Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: SIP vs. Skinny protocol
Skinny phone functionality is 'richer' than SIP phone functionality. First off, on a skinny phone, in hands free mode, you can start dialling and the phone will automatically go off hook. Sip requires you to manually hit the speaker button, hit new call, or pickup the phone before dialling. (One extra confusing key stroke I have a hard time getting over). I don't think SIP will work with the expansion modules on a 7960. Those are a few things I've found. On Asterisk there is a chan_skinny and a chan_sccp available for skinny based phones. Perhaps as more Cisco phones get used with *, more features will get implemented so they respond in a fashion very similar to a Callmanager installation. Maybe Cisco is already doing that in their labs? That would be cool. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Pauly Sent: December 23, 2003 12:52 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] OT: SIP vs. Skinny protocol I assume there are several people on this list that have Cisco Call Manager implementations under their belt We are beginning a call manager implementation and the first question I asked Cisco was, should we use SIP or Skinny. Cisco is pushing me towards Skinny, saying that I will lose some functionality with SIP. They also say that most of their customers implement skinny. I see two obvious benefits to using SIP: 1. I can get cheaper phones that run SIP, altough Cisco just came out with a 7902G for $130 US. 2. It's an open protocol and is more likely to survive long-term. What functionality do I lose by going with Skinny? Will Cisco eventually go with SIP only and I'll have to convert anyway? Any other pluses or minuses? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_skinny Feature set Development
Hello ... I'm working with SCCP only phones (ie, Cisco 7910s) and happened to notice that the chan_skinny driver seems to be missing some significant features. Most, if not all, button features (STIMULUS messages) are not implemented and callwaiting crashes the phone. Has there been much development with this driver not part of the standard CVS tree, or is there a diferent driver that is more complete? I tried chan_sccp, but it segfaults soon after the phone registers. Any help would be appreciated. Thanks! Lion Templin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: SIP vs. Skinny protocol
At 11:10 AM 12/24/2003, you wrote: Skinny phone functionality is 'richer' than SIP phone functionality. First off, on a skinny phone, in hands free mode, you can start dialling and the phone will automatically go off hook. Sip requires you to manually hit the speaker button, hit new call, or pickup the phone before dialling. (One extra confusing key stroke I have a hard time getting over). Um, that's a feature of the phone, not of the SIP protocol. My SNOM 200 lets me dial before picking up the handset no problem. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip phones on the same extension?
On Wednesday 24 December 2003 13:06, Brian Buhrow wrote: Hello. I'm a new Asterisk user, but I'm impressed with the flexibility and versatility of Asterisk, and am moving quickly to adopt it's main-line use in our company. Hopefully, you'll be hearing more from me as the project moves forward. Right now, though, I have a question about SIP peer registration. Right now, for our SIP-based phone,s, we're using the Sip Express Router product, which accepts sip registration requests and lets us route calls to any of the phones which register with SER. I am a semi-nomatic user, and can work at any of three different locations. Right now, my phones all sign up with SER, and register with the same telephone number. When someone dials that number, all three phones ring, and which ever one gets answered first, gets the call. When I tried to do this with Asterisk, sources from the cvs repository as of 12/18/2003, sip show peers only showed the most recent registration. This lead me to believe that if I dialed the number, only the most recently registered phone would ring. I was able to work around the problem by defining an umbrella extension which rings all three phones at the same time, but I'd like to have a way of dynamically adding phones to a given extension without having to necessarily rewrite the extensions.conf file, and I'd like calls from these extensions to show up from the master extension that folks should use to reach me. I imagine I could do something with pickup groups, but my understanding is that it is not true that all phones in a pickup group will necessarily ring just because they're a member of a given pickup group. The phones on this particular extension are many miles from each other, so one couldn't hear the other phone ring. Another work around is to put Asterisk behind SER, but this seems overly complicated, and I want to make sure that Asterisk doesn't do what I want before I pursue that path. Any suggestions on how to have multiple phones register with the same number in Asterisk? In sip.conf: [phone1] type=peer host=dynamic [phone2] type=peer host=dynamic [phone3] type=peer host=dynamic in extensions.conf: [default] exten = 0,1,Dial(SIP/phone1SIP/phone2SIP/phone3,30,T) -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] time to build an open phone?
Open software seems to work. Why don't we try it with hardware. 1. pick an embedded processor. It should have a nice linux gui support (like x jtag debugger). 2. pick a linux based cad system we all have easy access to and place schematics under cvs. 3. pick some type of gpio or serial interface for keyboard/display. 4. pick some basic functionality. 5. code it up. A stripped down *. Let everyone do their own thing with the expensive part. Tooling/packaging. We could let Digium be the distributor, so they are not left out of the loop. A board set would be offered with NO support. If Digium wants no part of it, we just build them on our own for our own use or sell them on ebay. What we would provide is schematics and source code. Everyone can take this to their favorite fab house and crank em out. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Merry Christmas and Happy New Year from XVOIP
Dear All, On behalf of XVOIP, LLC/Stealth Telecommunications, WISH YOU A MERRY CHRISTMAS AND A HAPPY PROSPEROUS NEW YEAR. Thanks to everyone for such great place as Asterisk community, for all your answers, suggestions, time, examples, help. We plan to support Asterisk project and in 2004 we will be launching new projects, which will include: 1700/777 access number from PSTN to IAXTEL in US/Canada (free access), Personal LCR (least cost routing engine), IAX/XVOIP Exchange. US DIDs in 40 US States and Ontario Province, Canada. All announcements will be posted before New Year to [EMAIL PROTECTED] list and asterisk-users mailing lists. Asterisk mailing-list members will be receiving special bonuses and will have priority on our network. Asterisk is an International community and let me try to congratulate you in your own language (See below ;-)). MERRY CHRISTMAS AND HAPPY NEW YEAR! Alexander Kandelaki Afrikaans - Geseknde Kersfees en 'n gelukkige nuwe jaar Argentine - Feliz Navidad y Feliz Año Nuevo Bohemian - Vesele Vanoce Brazilian - Boas Festas e Feliz Ano Novo Bulgarian - Vesela Koleda i chestita nova godina! Catalan - Bon Nadal i un Bon Any Nou! Chinese - Sing Dan Fae Lok. Gung Hai Fat Choi (Cantonese) Chinese - Shen Dan Kuai Le Xin Nian Yu Kuai (Mandarin) Chinese - Shen tan jie kuai le. Hsin Nien Kuaile Croatian - Sretan Bozic Czech - Stastne a vesele vanoce a stastny novy rok! Danish - Glaedelig Jul og godt nyter Dutch - Vrolijk Kerstfeest en een Gelukkig Nieuw Jaar Dutch - Prettige kerstdagen en een gelukkig nieuw jaar English - Merry Christmas and a Happy New Year Eskimo - (inupik) Jutdlime pivdluarit ukiortame pivdluaritlo! Esperanto - Felican Kristnaskon kaj Bonan Novjaron! Estonian - Rõõmusaid jõulupühi ja head uut aastat! Faeroese - Gledhilig jol og eydnurikt nyggjar! Filipinos - Maligayang Pasko Finnish - Hyvää joulua ja onnellista uutta vuotta! Flemish - Zalig Kerstfeest en Gelukkig nieuw jaar French - Joyeux Noel et Bonne Année! Scots Gaelic - Nollaig chridheil agus Bliadhna mhath yr! Galician - Bo Nadal German - Frohe Weihnachten und ein gl|ckliches Neues Jahr! Greek - Hronia polla kai eytyhismenos o kainourios hronos Greek - Hronia polla ke eftihismenos o kenourios hronos Hausa - Barka da Kirsimatikuma Barka da Sabuwar Shekara! Hawaian - Mele Kalikimaka ame Hauoli Makahiki Hou! Hungarian - Kellemes karacsonyi uennepeket es boldog ujevet! Icelandic - Gledhileg jsl og farsflt komandi ar! Indonesian - Selamat Hari Natal dan Selamat Tahun Baru! Iraqi - Idah Saidan Wa Sanah Jadidah Irish Gaelic - Nollaig Shona duit Irish Gaelic - Nollaig Shona Irish Gaelic - Nollaig faoi shean agus faoi shonas duit agus bliain nua faoi mhaise dhuit! Italian - Buon Natale e Felice Anno Nuovo! Japanese - Meri Kurisumasu soshite Akemashite Omedeto! Latin - Natale hilare et Annum Faustum! Latvian - Priecigus Ziemsvetkus un Laimigu Jaungadu! Lithuanian - Linksmu Kaledu Maltese - Nixtieklek Milied tajjeb u is-sena t-tabja! Modern Greek - Kala Christougenna kai evtichismenos o kainourios chronos! Norwegian - God Jul Og Godt Nytt Aar Pennsylvania German - En frehlicher Grischtdaag un en hallich Nei Yaahr! Polish - Vesowe Boze Narodzenie Polish - Wesolych Swiat i Szczesliwego Nowego Roku Portuguese - Boas Festas Portuguese - Feliz Natal e um Prospero Ano Novo Romanian - Craciun fericit si un an nou fericit Russian - S nastupaiushchim Novym godom i s Rozhdestvom Khristovym! Romanche - (sursilvan dialect): Legreivlas fiastas da Nadal e bien niev onn! Serbian - Hristos se rodi Slovakian - Sretan Bozic or Vesele vianoce Slovak - Vesele Vianoce i na zdravie v novom roku! Slovenian - Vesele bozicne praznike in srecno novo leto Spanish - Feliz Navidad y Próspero Año Nuevo Swedish - God Jul Och Ett Gott Nytt Ar Thai - Suk san wan Christmas Thai - Suk san wan pee mai - Happy New Year Trukeese - (Micronesian) Neekiriisimas annim oo iyer seefe feyiyeech! Turkish - Noeliniz kutlu olsun ve yeni yilinis kutlu olsun! Turkish - Noeliniz Ve Yeni Yiliniz Kutlu Olsun Ukrainian - Srozhdestvom Kristovym Ukrainan - Z novym rokom i s rizdvom Hrystovym! Ukrainan - Khrystos Rodevsia Vietnamese - Chuc mung nam moi va Giang Sinh vui ve Welsh - Nadolig Llawen a Blwyddyn Newydd Da! Yoruba - E ku odun, e ku iye'dun! image001.gif
RE: [Asterisk-Users] Grandstream Quality Survey.... :P
My phone's booted up and registered OK but a strange thing noticed on the tftp uploads. bootloader.bin bt.bin voc.bin html.bin vp.bin ht.bin The first phone uploaded the first four bin files. The second phone uploaded the first five bin files. Neither phone uploaded the ht.bin file. Both phones asjed for a file called cfg.txt, which isn't there. Any thoughts on why the phones uploaded some but not all files? Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brian West Sent: 24 December 2003 15:32 To: Asterisk List Subject: Re: [Asterisk-Users] Grandstream Quality Survey :P Yes when you upgrade to beta code you may have to reboot 3 times for the phone to function properly. Then cross your fingers that the phone will accually register with * once you do that. bkw On Wed, 24 Dec 2003, Dave Cotton wrote: On Wed, 2003-12-24 at 15:50, WipeOut wrote: I just loaded the b13p4.30.zip firmware and now I am not able to log into the GS admin interface.. anyone else having this problem? Yep been there. Panicked, rebooted the phone and it responded as normal. I just tried it again because of your question and had to reboot again to get in. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] time to build an open phone?
I had been thinking of doing this, but lack the electronics expertise to do such a thing. I basically need phones that look like trading turrets, so I can sneak them into this one trading firm. Good idea, let's see if there's any traction. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob Knight Sent: Wednesday, December 24, 2003 1:30 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] time to build an open phone? Open software seems to work. Why don't we try it with hardware. 1. pick an embedded processor. It should have a nice linux gui support (like x jtag debugger). 2. pick a linux based cad system we all have easy access to and place schematics under cvs. 3. pick some type of gpio or serial interface for keyboard/display. 4. pick some basic functionality. 5. code it up. A stripped down *. Let everyone do their own thing with the expensive part. Tooling/packaging. We could let Digium be the distributor, so they are not left out of the loop. A board set would be offered with NO support. If Digium wants no part of it, we just build them on our own for our own use or sell them on ebay. What we would provide is schematics and source code. Everyone can take this to their favorite fab house and crank em out. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk- users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unlocking Vonage ATA 186
In the process of investigating a Cisco ATA 186 that was locked by Vonage, I found that you can still unlock the device yourself. But there's a catch. The device's design has a great plus: a DIP32 *socketed* SST28SF040A flash chip. I found an 8 digit unlock code at 0x03FA71-0x03FA78. I do not know if that is a standard location. If you have the equipment, you're in luck. But IMHO, the $15 fee is more than reasonable .. and certainly less than what it would cost to get a device to read/write these flash chips. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Reversing a Firmware Upgrade
My Grandstream phone seems quite happy to accept a new firmware, but having tried the latest beta firmware, which I am unhappy with I want to update with an older version. How do I do this? Thanks, Michael -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unlocking Vonage ATA 186
I asked Vonage about unlocking it and they refused to. They don't offer an unlock service for $15. Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lion Templin Sent: Wednesday, December 24, 2003 2:53 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Unlocking Vonage ATA 186 In the process of investigating a Cisco ATA 186 that was locked by Vonage, I found that you can still unlock the device yourself. But there's a catch. The device's design has a great plus: a DIP32 *socketed* SST28SF040A flash chip. I found an 8 digit unlock code at 0x03FA71-0x03FA78. I do not know if that is a standard location. If you have the equipment, you're in luck. But IMHO, the $15 fee is more than reasonable .. and certainly less than what it would cost to get a device to read/write these flash chips. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FWD problems
--On Wednesday, December 24, 2003 6:35 pm +0100 Arnold Ligtvoet [EMAIL PROTECTED] wrote: Read the fwd announcement. Jeff Pulver mentioned the fact that * users cannot use the free holiday calls, since FWD cannot be sure that * is not being used by more than 1 user at the same time. Where in this announcement: On Free World Dialup, go ahead and dial: *1 (area code) Number. We have arranged to pick up the costs to allow members of the Free World Dialup community to place calls into the US and Canada for Free during the 2003 holiday season. While the offical press release will follow later today or tomorrow, you can help out in the beta-trials of this holiday gift today. Feel free to share the holiday spirit and cheer. :-) .. does it say * cannot be used? Remember, I tried this a week ago and got the this service is available to FWD members only message. Pulver posted the message mentioning the restriction on 21 December - I've been waiting since December 18 for a reply to my original report of a problem. Still, there seems to be a you get what you pay for theme to many of today's posts and this clearly applies to support on FWD. Naybe we should remove the signature from * that enables FWD to identify * systems :-) Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] registration problem
Hi, Why do I get registration refused errors with Asterisk and voip providers? I did everything correctly and every provider I signed up with gives me that error: Dec 24 15:30:13 NOTICE[-1254995024]: File chan_iax.c, Line 3955 (socket_read): Registration of 'in-STn46BoD89' rejected: Registration Refused Dec 24 15:30:13 NOTICE[-1298793552]: File chan_iax2.c, Line 4389 (socket_read): Registration of 'in-STn46BoD89' rejected: Registration Refused Dec 24 15:30:13 NOTICE[-1298793552]: File chan_iax2.c, Line 4389 (socket_read): Registration of 'mzmahoney' rejected: Registration Refused Thank You, Matt
RE: [Asterisk-Users] FWD problems
Still, there seems to be a you get what you pay for theme to many of today's posts and this clearly applies to support on FWD. Naybe we should remove the signature from * that enables FWD to identify * systems :-) That certainly seems the case for today's theme... It is certainly the right of any company or person to define the rules of their service. Since I don't pay for either Asterisk or FWD then I appreciate the service that is provided and try not to crusify them when things don't go right. This entire VoIP is still rather experimental. If I want guaranteed service then I'll pay some provider for it... THEN.. and only then will a service level be expected. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: SIP vs. Skinny protocol
Skinny phone functionality is 'richer' than SIP phone functionality. First off, on a skinny phone, in hands free mode, you can start dialling and the phone will automatically go off hook. Sip requires you to manually hit the speaker button, hit new call, or pickup the phone before dialling. (One extra confusing key stroke I have a hard time getting over). This is not a sip issue, it's a phone funcionality... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: SIP vs. Skinny protocol
Ray Burkholder wrote: Skinny phone functionality is 'richer' than SIP phone functionality. First Skinny *functionality* seems to be 'richer', but it's implementation in * is woefully under-functional. Regardless of individual phone feature sets, SIP is far better implemented in * than skinny. Most features of the 7910s I have don't even have supporting code written. -- -- = lion is Lion J Templin [EMAIL PROTECTED] = = 612-605-3613 x3001 FWD 94117 = ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FWD problems
--On Wednesday, December 24, 2003 10:06 pm +0100 rnc Info Lists [EMAIL PROTECTED] wrote: Still, there seems to be a you get what you pay for theme to many of today's posts and this clearly applies to support on FWD. Naybe we should remove the signature from * that enables FWD to identify * systems :-) That certainly seems the case for today's theme... It is certainly the right of any company or person to define the rules of their service. Since I don't pay for either Asterisk or FWD then I appreciate the service that is provided and try not to crusify them when things don't go right. This entire VoIP is still rather experimental. If I want guaranteed service then I'll pay some provider for it... THEN.. and only then will a service level be expected. That's fair comment but I think FWD should have put a correct message on their system for asterisk users. It wouldn't have taken much effort. FWD and * complement each other and should benefit from each other's success. Indeed * is cited on the FWD web site and mentioned by Jeff Pulver at his VON events. It seems a little unfortunate that FWD is assuming all * systems are a front for hundreds of users and banning them. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Capi Dial outgoing msn?
Hi Patrick,
Citeren Patrick [EMAIL PROTECTED]:
I am trying to get Capi Dial to use a specific outgoing msn. I can't get
it to work. If I make a test call to 0703241494 (same isdn line, just
one of the other numbers) I don't get CLID at all. Any ideas?
; use 0703241432 as outgoing msn
exten = _070.,1,Dial(CAPI/@0703241432:${EXTEN}|30|r)
Its the wonderfull world of KPN. You need to drop the 0. This works for me:
exten = _XXX,1,Dial(CAPI/534281234:b${EXTEN})
--
Met vriendelijke groet,
Florian Overkamp
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Re: [Asterisk-Users] Grandstream Quality Survey.... :P
We bought 50 of these phones and deployed them at customer sites. But after 4 months of operation we have decided that they are completely unfit for our use. The have many bugs. The worst one is the one where the phone stops registering, others include: web page dies, numerous break in the SIP protocol, breaks in the UDP stack, problems with STUN operation, charging for GAPS!, etc... As a service provider the cost of hiring extra people to attend the increased workload of technical support to customers, far outweighs the $65 price. On the other hand we have most of our customers using the ATA186 and even though it costs 2X of what the Grandsrtream does, it is cheaper for us to support. And add to the fact that ATA186 customers are extremely happy with our service but Grandstream ones are at times infuriated. These phones might work fine for a small office but do not scale well with the requirements of a service provider. We are now pulling them all out of our network and will be dumping them on eBay. We are beginning our tests with the SPA2000. Andres. http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FWD problems
On Wed, 24 Dec 2003, Iain Stevenson wrote: I have exactly this problem and posted a bug report to FWD about a week ago - no response yet. It's bizarre that FWD recognises you to dial another user but not to call outside their network. Sounds more like a FWD problem than a * problem to me. Suspect your INVITE into FWD isn't authenticated so FWD thinks of you as a foreigner. Perhaps a sip debug will help see what is happening. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Reversing a Firmware Upgrade
Michael, A reply I received from Grandstream. Depending on your firmware version. Firmware family 1.0.4.x is not interchangeable with 1.0.3.x and therefore cannot downgrade back. What is the current firmware version and what version do you want to roll back to? Regards, Richard Huang Grandstream Technical Support Hope this helps. Regards Dave Tel: - +44 (0) 709 215 4039 SipPhone :- 1 747 669 1957 Iaxtel: - 1 700 818 8820 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael T Farnworth Sent: 24 December 2003 20:05 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Reversing a Firmware Upgrade My Grandstream phone seems quite happy to accept a new firmware, but having tried the latest beta firmware, which I am unhappy with I want to update with an older version. How do I do this? Thanks, Michael -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G729 troubles
Hello,
I've successfully installed Asterisk from last CVS and configured it
for using with DLINK-DG104S as mgcp CPE and PGW2200 as external sip
server.
All are work fine at G711 codecs, but then I disable all codecs except
g729 some calls failed (Not all calls. Some calls passed at g729
succesfully).
All my devices configred to use only g729 and I don't see other codecs
at mgcp or sip messages, but I see strange string at asterisks log:
NOTICE[196633]: File rtp.c, Line 418 (ast_rtp_read): Unknown RTP codec
123 received
NOTICE[196633]: File channel.c, Line 1478 (ast_set_read_format): Unable
to find a path from ALAW to G729A
NOTICE[196633]: File channel.c, Line 1448 (ast_set_write_format): Unable
to find a path from G729A to ALAW
WARNING[196633]: File chan_mgcp.c, Line 757 (mgcp_write): Asked to
transmit frame type 8, while native formats is 256 (read/write =
256/256)
WARNING[196633]: File app_dial.c, Line 279 (wait_for_answer): Unable to
forward frame
I find similary posts at Asteris-Users mailing list, but don't find how
to resolve this trouble. Is this a bug or some misconfiguration at my
configs ?
sip.conf:
[general]
port = 5060
bindaddr = 0.0.0.0
context = local
disallow = all
allow = g729
mgcp.conf
[general]
port = 2427
bindaddr = 0.0.0.0
disallow = all
allow = g729
[DLINK]
context=local
host=Y.Y.Y.Y
threewaycalling=yes
transfer=yes
line = aaln/1
line = aaln/2
line = aaln/3
line = aaln/4
extension.conf
[local]
ignorepat = 9
exten = _9XXX,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]
Some logs from Asterisk:
First mgcp CRCX after hang up:
Posting Request:
CRCX 323 aaln/[EMAIL PROTECTED] MGCP 1.0
v=0
o=root 23577 23577 IN IP4 X.X.X.X
s=session
c=IN IP4 X.X.X.X
t=0 0
m=audio 14548 RTP/AVP 18
a=rtpmap:18 G729/8000
After that I enter phone number and sent call to sip server:
-- Executing Dial(MGCP/aaln/[EMAIL PROTECTED], SIP/[EMAIL PROTECTED])
in new stack
INVITE sip:[EMAIL PROTECTED] SIP/2.0
skip
v=0
o=root 16078 16078 IN IP4 X.X.X.X
s=session
c=IN IP4 X.X.X.X
t=0 0
m=audio 18480 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Then I receive reply from SIP server:
Sip read:
SIP/2.0 100 Trying
skip
Sip read:
SIP/2.0 183 Session Progress
skip
v=0
o=- 0 0 IN IP4 Z.Z.Z.Z
s=-
c=IN IP4 Z.Z.Z.Z
t=0 0
m=audio 49640 RTP/AVP 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-sqn: 0
a=X-cap: 1 image udptl t38
a=sqn: 0
a=cdsc: 1 image udptl t38
After this message sometimes Asterisk make error message at log and drop
call:
-- SIP/IP.IP.IP.IP-b782 is making progress passing it to
MGCP/aaln/[EMAIL PROTECTED]
srv-5*CLI NOTICE[196633]: File rtp.c, Line 418 (ast_rtp_read): Unknown
RTP codec 123 received
NOTICE[196633]: File channel.c, Line 1478 (ast_set_read_format): Unable
to find a path from ALAW to G729A
NOTICE[196633]: File channel.c, Line 1448 (ast_set_write_format): Unable
to find a path from G729A to ALAW
WARNING[196633]: File chan_mgcp.c, Line 757 (mgcp_write): Asked to
transmit frame type 8, while native formats is 256 (read/write =
256/256)
WARNING[196633]: File app_dial.c, Line 279 (wait_for_answer): Unable to
forward frame
Reliably Transmitting:
CANCEL sip:[EMAIL PROTECTED]:5060 SIP/2.0
Sip read:
SIP/2.0 487 Request Cancelled
--
Antonio
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Re: [Asterisk-Users] G729 troubles
Have you bought G.729a from Digium which cost $10/channel?
At 02:04 25/12/03 +0300, you wrote:
Hello,
I've successfully installed Asterisk from last CVS and configured it
for using with DLINK-DG104S as mgcp CPE and PGW2200 as external sip
server.
All are work fine at G711 codecs, but then I disable all codecs except
g729 some calls failed (Not all calls. Some calls passed at g729
succesfully).
All my devices configred to use only g729 and I don't see other codecs
at mgcp or sip messages, but I see strange string at asterisks log:
NOTICE[196633]: File rtp.c, Line 418 (ast_rtp_read): Unknown RTP codec
123 received
NOTICE[196633]: File channel.c, Line 1478 (ast_set_read_format): Unable
to find a path from ALAW to G729A
NOTICE[196633]: File channel.c, Line 1448 (ast_set_write_format): Unable
to find a path from G729A to ALAW
WARNING[196633]: File chan_mgcp.c, Line 757 (mgcp_write): Asked to
transmit frame type 8, while native formats is 256 (read/write =
256/256)
WARNING[196633]: File app_dial.c, Line 279 (wait_for_answer): Unable to
forward frame
I find similary posts at Asteris-Users mailing list, but don't find how
to resolve this trouble. Is this a bug or some misconfiguration at my
configs ?
sip.conf:
[general]
port = 5060
bindaddr = 0.0.0.0
context = local
disallow = all
allow = g729
mgcp.conf
[general]
port = 2427
bindaddr = 0.0.0.0
disallow = all
allow = g729
[DLINK]
context=local
host=Y.Y.Y.Y
threewaycalling=yes
transfer=yes
line = aaln/1
line = aaln/2
line = aaln/3
line = aaln/4
extension.conf
[local]
ignorepat = 9
exten = _9XXX,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]
Some logs from Asterisk:
First mgcp CRCX after hang up:
Posting Request:
CRCX 323 aaln/[EMAIL PROTECTED] MGCP 1.0
v=0
o=root 23577 23577 IN IP4 X.X.X.X
s=session
c=IN IP4 X.X.X.X
t=0 0
m=audio 14548 RTP/AVP 18
a=rtpmap:18 G729/8000
After that I enter phone number and sent call to sip server:
-- Executing Dial(MGCP/aaln/[EMAIL PROTECTED], SIP/[EMAIL PROTECTED])
in new stack
INVITE sip:[EMAIL PROTECTED] SIP/2.0
skip
v=0
o=root 16078 16078 IN IP4 X.X.X.X
s=session
c=IN IP4 X.X.X.X
t=0 0
m=audio 18480 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Then I receive reply from SIP server:
Sip read:
SIP/2.0 100 Trying
skip
Sip read:
SIP/2.0 183 Session Progress
skip
v=0
o=- 0 0 IN IP4 Z.Z.Z.Z
s=-
c=IN IP4 Z.Z.Z.Z
t=0 0
m=audio 49640 RTP/AVP 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-sqn: 0
a=X-cap: 1 image udptl t38
a=sqn: 0
a=cdsc: 1 image udptl t38
After this message sometimes Asterisk make error message at log and drop
call:
-- SIP/IP.IP.IP.IP-b782 is making progress passing it to
MGCP/aaln/[EMAIL PROTECTED]
srv-5*CLI NOTICE[196633]: File rtp.c, Line 418 (ast_rtp_read): Unknown
RTP codec 123 received
NOTICE[196633]: File channel.c, Line 1478 (ast_set_read_format): Unable
to find a path from ALAW to G729A
NOTICE[196633]: File channel.c, Line 1448 (ast_set_write_format): Unable
to find a path from G729A to ALAW
WARNING[196633]: File chan_mgcp.c, Line 757 (mgcp_write): Asked to
transmit frame type 8, while native formats is 256 (read/write =
256/256)
WARNING[196633]: File app_dial.c, Line 279 (wait_for_answer): Unable to
forward frame
Reliably Transmitting:
CANCEL sip:[EMAIL PROTECTED]:5060 SIP/2.0
Sip read:
SIP/2.0 487 Request Cancelled
--
Antonio
___
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Peter Brown
CEO
IP Telephonics
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G729 troubles
No, I did't bought any license from Digium. But as I say at my previous
post, only _some part_ of my g729 calls are failed !
I think if I need license for G729 at Asterisk then all of my calls must
to fails. Is it right ?
--
Antonio
-Original Message-
From: Peter Brown [mailto:[EMAIL PROTECTED]
Sent: Thursday, December 25, 2003 2:50 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] G729 troubles
Have you bought G.729a from Digium which cost $10/channel?
At 02:04 25/12/03 +0300, you wrote:
Hello,
I've successfully installed Asterisk from last CVS and
configured it
for using with DLINK-DG104S as mgcp CPE and PGW2200 as external sip
server.
All are work fine at G711 codecs, but then I disable all
codecs except
g729 some calls failed (Not all calls. Some calls passed at g729
succesfully).
All my devices configred to use only g729 and I don't see
other codecs
at mgcp or sip messages, but I see strange string at asterisks log:
NOTICE[196633]: File rtp.c, Line 418 (ast_rtp_read): Unknown
RTP codec
123 received
NOTICE[196633]: File channel.c, Line 1478
(ast_set_read_format): Unable
to find a path from ALAW to G729A
NOTICE[196633]: File channel.c, Line 1448 (ast_set_write_format):
Unable to find a path from G729A to ALAW
WARNING[196633]: File chan_mgcp.c, Line 757 (mgcp_write): Asked to
transmit frame type 8, while native formats is 256 (read/write =
256/256)
WARNING[196633]: File app_dial.c, Line 279
(wait_for_answer): Unable to
forward frame
I find similary posts at Asteris-Users mailing list, but
don't find how
to resolve this trouble. Is this a bug or some
misconfiguration at my
configs ?
sip.conf:
[general]
port = 5060
bindaddr = 0.0.0.0
context = local
disallow = all
allow = g729
mgcp.conf
[general]
port = 2427
bindaddr = 0.0.0.0
disallow = all
allow = g729
[DLINK]
context=local
host=Y.Y.Y.Y
threewaycalling=yes
transfer=yes
line = aaln/1
line = aaln/2
line = aaln/3
line = aaln/4
extension.conf
[local]
ignorepat = 9
exten = _9XXX,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]
Some logs from Asterisk:
First mgcp CRCX after hang up:
Posting Request:
CRCX 323 aaln/[EMAIL PROTECTED] MGCP 1.0
v=0
o=root 23577 23577 IN IP4 X.X.X.X
s=session
c=IN IP4 X.X.X.X
t=0 0
m=audio 14548 RTP/AVP 18
a=rtpmap:18 G729/8000
After that I enter phone number and sent call to sip server:
-- Executing Dial(MGCP/aaln/[EMAIL PROTECTED],
SIP/[EMAIL PROTECTED]) in new stack
INVITE sip:[EMAIL PROTECTED] SIP/2.0
skip
v=0
o=root 16078 16078 IN IP4 X.X.X.X
s=session
c=IN IP4 X.X.X.X
t=0 0
m=audio 18480 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Then I receive reply from SIP server:
Sip read:
SIP/2.0 100 Trying
skip
Sip read:
SIP/2.0 183 Session Progress
skip
v=0
o=- 0 0 IN IP4 Z.Z.Z.Z
s=-
c=IN IP4 Z.Z.Z.Z
t=0 0
m=audio 49640 RTP/AVP 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-sqn: 0
a=X-cap: 1 image udptl t38
a=sqn: 0
a=cdsc: 1 image udptl t38
After this message sometimes Asterisk make error message at log and
drop
call:
-- SIP/IP.IP.IP.IP-b782 is making progress passing it to
MGCP/aaln/[EMAIL PROTECTED]
srv-5*CLI NOTICE[196633]: File rtp.c, Line 418
(ast_rtp_read): Unknown
RTP codec 123 received
NOTICE[196633]: File channel.c, Line 1478
(ast_set_read_format): Unable
to find a path from ALAW to G729A
NOTICE[196633]: File channel.c, Line 1448 (ast_set_write_format):
Unable to find a path from G729A to ALAW
WARNING[196633]: File chan_mgcp.c, Line 757 (mgcp_write): Asked to
transmit frame type 8, while native formats is 256 (read/write =
256/256)
WARNING[196633]: File app_dial.c, Line 279
(wait_for_answer): Unable to
forward frame
Reliably Transmitting:
CANCEL sip:[EMAIL PROTECTED]:5060 SIP/2.0
Sip read:
SIP/2.0 487 Request Cancelled
--
Antonio
___
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Peter Brown
CEO
IP Telephonics
___
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___
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RE: [Asterisk-Users] G729 troubles
I'm going to take a stab at this, so someone correct me if I'm wrong! If
you're calling one g729 device from another, the call is actually handed off
without any decoding done, therefore the licensing isn't needed. If * has
to connect the g729 call to another format, then the licensing comes in to
play. And it could be that even though you've configured the disabling of
the codec at one location, it still is enabled elsewhere? Close? Anyone?
Sean
-Original Message-
From: Anton V Kirichenko [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 24, 2003 7:04 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] G729 troubles
No, I did't bought any license from Digium. But as I say at my previous
post, only _some part_ of my g729 calls are failed !
I think if I need license for G729 at Asterisk then all of my calls must
to fails. Is it right ?
--
Antonio
-Original Message-
From: Peter Brown [mailto:[EMAIL PROTECTED]
Sent: Thursday, December 25, 2003 2:50 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] G729 troubles
Have you bought G.729a from Digium which cost $10/channel?
At 02:04 25/12/03 +0300, you wrote:
Hello,
I've successfully installed Asterisk from last CVS and
configured it
for using with DLINK-DG104S as mgcp CPE and PGW2200 as external sip
server.
All are work fine at G711 codecs, but then I disable all
codecs except
g729 some calls failed (Not all calls. Some calls passed at g729
succesfully).
All my devices configred to use only g729 and I don't see
other codecs
at mgcp or sip messages, but I see strange string at asterisks log:
NOTICE[196633]: File rtp.c, Line 418 (ast_rtp_read): Unknown
RTP codec
123 received
NOTICE[196633]: File channel.c, Line 1478
(ast_set_read_format): Unable
to find a path from ALAW to G729A
NOTICE[196633]: File channel.c, Line 1448 (ast_set_write_format):
Unable to find a path from G729A to ALAW
WARNING[196633]: File chan_mgcp.c, Line 757 (mgcp_write): Asked to
transmit frame type 8, while native formats is 256 (read/write =
256/256)
WARNING[196633]: File app_dial.c, Line 279
(wait_for_answer): Unable to
forward frame
I find similary posts at Asteris-Users mailing list, but
don't find how
to resolve this trouble. Is this a bug or some
misconfiguration at my
configs ?
sip.conf:
[general]
port = 5060
bindaddr = 0.0.0.0
context = local
disallow = all
allow = g729
mgcp.conf
[general]
port = 2427
bindaddr = 0.0.0.0
disallow = all
allow = g729
[DLINK]
context=local
host=Y.Y.Y.Y
threewaycalling=yes
transfer=yes
line = aaln/1
line = aaln/2
line = aaln/3
line = aaln/4
extension.conf
[local]
ignorepat = 9
exten = _9XXX,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]
Some logs from Asterisk:
First mgcp CRCX after hang up:
Posting Request:
CRCX 323 aaln/[EMAIL PROTECTED] MGCP 1.0
v=0
o=root 23577 23577 IN IP4 X.X.X.X
s=session
c=IN IP4 X.X.X.X
t=0 0
m=audio 14548 RTP/AVP 18
a=rtpmap:18 G729/8000
After that I enter phone number and sent call to sip server:
-- Executing Dial(MGCP/aaln/[EMAIL PROTECTED],
SIP/[EMAIL PROTECTED]) in new stack
INVITE sip:[EMAIL PROTECTED] SIP/2.0
skip
v=0
o=root 16078 16078 IN IP4 X.X.X.X
s=session
c=IN IP4 X.X.X.X
t=0 0
m=audio 18480 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Then I receive reply from SIP server:
Sip read:
SIP/2.0 100 Trying
skip
Sip read:
SIP/2.0 183 Session Progress
skip
v=0
o=- 0 0 IN IP4 Z.Z.Z.Z
s=-
c=IN IP4 Z.Z.Z.Z
t=0 0
m=audio 49640 RTP/AVP 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-sqn: 0
a=X-cap: 1 image udptl t38
a=sqn: 0
a=cdsc: 1 image udptl t38
After this message sometimes Asterisk make error message at log and
drop
call:
-- SIP/IP.IP.IP.IP-b782 is making progress passing it to
MGCP/aaln/[EMAIL PROTECTED]
srv-5*CLI NOTICE[196633]: File rtp.c, Line 418
(ast_rtp_read): Unknown
RTP codec 123 received
NOTICE[196633]: File channel.c, Line 1478
(ast_set_read_format): Unable
to find a path from ALAW to G729A
NOTICE[196633]: File channel.c, Line 1448 (ast_set_write_format):
Unable to find a path from G729A to ALAW
WARNING[196633]: File chan_mgcp.c, Line 757 (mgcp_write): Asked to
transmit frame type 8, while native formats is 256 (read/write =
256/256)
WARNING[196633]: File app_dial.c, Line 279
(wait_for_answer): Unable to
forward frame
Reliably Transmitting:
CANCEL sip:[EMAIL PROTECTED]:5060 SIP/2.0
Sip read:
SIP/2.0 487 Request Cancelled
--
Antonio
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Peter Brown
CEO
IP Telephonics
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Re: [Asterisk-Users] Grandstream Quality Survey.... :P
Where can you get a cisco 7905 with a SIP license and power supply for $99? Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 24, 2003 5:30 AM Subject: Re: [Asterisk-Users] Grandstream Quality Survey :P Cisco 7905's are damn fine phones for 99 bucks and they blow the grandstream away... bkw On Wed, 24 Dec 2003, Cameron Palmer wrote: It is unfortunate that Cisco is so damned expensive. $300 is only the tip of the iceberg if you go the official route... You still haven't paid for their ongoing maintenance. They should really consider selling their phones at a better price. cameron. On Wed, 24 Dec 2003, Robert Hajime Lanning wrote: So, you can get a really good analog phone for $65, then you mention and use an ata... what does this ATA cost? $65 for the complete set is what I pay for. At that price, I expect an issue here and there. It is still getting the bugs worked out. I don't have the money to buy $300 Cisco phones. quote who=Miguel Cavazos They are $65 yes, but you can get the best analog phones on the market for that price and use an ata. If GS could give the information for people on asterisk to develop iax this $65 phone could be even better than most of the phones in the market more features less buggy and cheaper than all the other sip phones out there ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G729 troubles
In my case I see only g729 codec request from CPE (see mgcp CRCX) and
only g729 from PGW2200 (see debug of sip messages) and I don't need and
transcoding from one codec format to another codec format.
Could you expain to me why asterisk starts transcoding process from
g729 to alaw ?
--
antonio
-Original Message-
From: Sean Cheesman [mailto:[EMAIL PROTECTED]
Sent: Thursday, December 25, 2003 3:34 AM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] G729 troubles
I'm going to take a stab at this, so someone correct me if
I'm wrong! If you're calling one g729 device from another,
the call is actually handed off without any decoding done,
therefore the licensing isn't needed. If * has to connect
the g729 call to another format, then the licensing comes in
to play. And it could be that even though you've configured
the disabling of the codec at one location, it still is
enabled elsewhere? Close? Anyone?
Sean
-Original Message-
From: Anton V Kirichenko [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 24, 2003 7:04 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] G729 troubles
No, I did't bought any license from Digium. But as I say at
my previous post, only _some part_ of my g729 calls are failed !
I think if I need license for G729 at Asterisk then all of my
calls must to fails. Is it right ?
--
Antonio
-Original Message-
From: Peter Brown [mailto:[EMAIL PROTECTED]
Sent: Thursday, December 25, 2003 2:50 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] G729 troubles
Have you bought G.729a from Digium which cost $10/channel?
At 02:04 25/12/03 +0300, you wrote:
Hello,
I've successfully installed Asterisk from last CVS and
configured it
for using with DLINK-DG104S as mgcp CPE and PGW2200 as
external sip
server.
All are work fine at G711 codecs, but then I disable all
codecs except
g729 some calls failed (Not all calls. Some calls passed at g729
succesfully).
All my devices configred to use only g729 and I don't see
other codecs
at mgcp or sip messages, but I see strange string at
asterisks log:
NOTICE[196633]: File rtp.c, Line 418 (ast_rtp_read): Unknown
RTP codec
123 received
NOTICE[196633]: File channel.c, Line 1478
(ast_set_read_format): Unable
to find a path from ALAW to G729A
NOTICE[196633]: File channel.c, Line 1448 (ast_set_write_format):
Unable to find a path from G729A to ALAW
WARNING[196633]: File chan_mgcp.c, Line 757 (mgcp_write): Asked to
transmit frame type 8, while native formats is 256 (read/write =
256/256)
WARNING[196633]: File app_dial.c, Line 279
(wait_for_answer): Unable to
forward frame
I find similary posts at Asteris-Users mailing list, but
don't find how
to resolve this trouble. Is this a bug or some
misconfiguration at my
configs ?
sip.conf:
[general]
port = 5060
bindaddr = 0.0.0.0
context = local
disallow = all
allow = g729
mgcp.conf
[general]
port = 2427
bindaddr = 0.0.0.0
disallow = all
allow = g729
[DLINK]
context=local
host=Y.Y.Y.Y
threewaycalling=yes
transfer=yes
line = aaln/1
line = aaln/2
line = aaln/3
line = aaln/4
extension.conf
[local]
ignorepat = 9
exten = _9XXX,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]
Some logs from Asterisk:
First mgcp CRCX after hang up:
Posting Request:
CRCX 323 aaln/[EMAIL PROTECTED] MGCP 1.0
v=0
o=root 23577 23577 IN IP4 X.X.X.X
s=session
c=IN IP4 X.X.X.X
t=0 0
m=audio 14548 RTP/AVP 18
a=rtpmap:18 G729/8000
After that I enter phone number and sent call to sip server:
-- Executing Dial(MGCP/aaln/[EMAIL PROTECTED],
SIP/[EMAIL PROTECTED]) in new stack
INVITE sip:[EMAIL PROTECTED] SIP/2.0 skip v=0 o=root
16078 16078
IN IP4 X.X.X.X s=session c=IN IP4 X.X.X.X t=0 0 m=audio
18480 RTP/AVP
18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Then I receive reply from SIP server:
Sip read:
SIP/2.0 100 Trying
skip
Sip read:
SIP/2.0 183 Session Progress
skip
v=0
o=- 0 0 IN IP4 Z.Z.Z.Z
s=-
c=IN IP4 Z.Z.Z.Z
t=0 0
m=audio 49640 RTP/AVP 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-sqn: 0
a=X-cap: 1 image udptl t38
a=sqn: 0
a=cdsc: 1 image udptl t38
After this message sometimes Asterisk make error message
at log and
drop
call:
-- SIP/IP.IP.IP.IP-b782 is making progress passing it to
MGCP/aaln/[EMAIL PROTECTED]
srv-5*CLI NOTICE[196633]: File rtp.c, Line 418
(ast_rtp_read): Unknown
RTP codec 123 received
NOTICE[196633]: File channel.c, Line 1478
(ast_set_read_format): Unable
to find a path from ALAW to G729A
NOTICE[196633]: File channel.c, Line 1448 (ast_set_write_format):
Unable to find a path from G729A to ALAW
WARNING[196633]: File chan_mgcp.c, Line 757 (mgcp_write): Asked to
transmit frame type 8, while native
[Asterisk-Users] Encryption
Hi, Does asterisk support any kind of voice encryption? Matt
Re: [Asterisk-Users] time to build an open phone?
Interesting! Surely it would be another greate project. Happy christmas! - Original Message - From: Bob Knight [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 24, 2003 11:30 AM Subject: [Asterisk-Users] time to build an open phone? Open software seems to work. Why don't we try it with hardware. 1. pick an embedded processor. It should have a nice linux gui support (like x jtag debugger). 2. pick a linux based cad system we all have easy access to and place schematics under cvs. 3. pick some type of gpio or serial interface for keyboard/display. 4. pick some basic functionality. 5. code it up. A stripped down *. Let everyone do their own thing with the expensive part. Tooling/packaging. We could let Digium be the distributor, so they are not left out of the loop. A board set would be offered with NO support. If Digium wants no part of it, we just build them on our own for our own use or sell them on ebay. What we would provide is schematics and source code. Everyone can take this to their favorite fab house and crank em out. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100p problem
Title: X100p problem I am having a problem with the x100p cards. It doesn't matter whether the card is in the machine or not, all I get is a busy signal when calling. The Asterisk box doesn't give me any errors and doesn't show that any call is coming through. I removed the cards from the machine completely and they still give busy signal when dialed. Any ideas? I must say that after dealing with the ordering process with Digium, and now the seemingly broken cards, I have to say that I completely frustrated and unhappy with deciding to go with digium. I think that Asterisk is probably very cool, and will do what I want, but it took three weeks to get my cards and the people at digium won't email to save their lives. Anyway, please help with the card problem as I feel that I am out another week and this was supposed to be running last week Thanks Sean Garland
RE: [Asterisk-Users] X100p problem
Hi Sean- I've had pretty good experience with Digium boards - all of mine have been shipped quickly, and all have worked upon arrival. Don't have experience with the X100P, only the quad T1/E1 boards. You didn't provide much information about what you tried already. Have you got /etc/zaptel.conf and /etc/asterisk/zapata.conf set up? Have you edited your extensions.conf file appropriately, do you have green lights on the card, etc. If you ask more specific questions, someone may be able to help you. I agree with you that Digium is a bit weak on their support procedures - the people are good to work with and helpful, but often its hard to get their attention. I say this in a public forum with the positive hope that perhaps they can invent some kind of priority system to at least support people like us who actually buy things and help their bottom line. Some updates to the board documentation would be helpful too. Anyway, tell us more and maybe we can help. Cheers Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: [EMAIL PROTECTED] URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Garland Sent: Thursday, December 25, 2003 3:31 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] X100p problem I am having a problem with the x100p cards. It doesn't matter whether the card is in the machine or not, all I get is a busy signal when calling. The Asterisk box doesn't give me any errors and doesn't show that any call is coming through. I removed the cards from the machine completely and they still give busy signal when dialed. Any ideas? I must say that after dealing with the ordering process with Digium, and now the seemingly broken cards, I have to say that I completely frustrated and unhappy with deciding to go with digium. I think that Asterisk is probably very cool, and will do what I want, but it took three weeks to get my cards and the people at digium won't email to save their lives. Anyway, please help with the card problem as I feel that I am out another week and this was supposed to be running last week. Thanks Sean Garland ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] when * start at bootup chan_h323 fails to load
Jeremy, Ok, that worked. Thanks for your help, really appreciate it. Let me copy this to the list, someone will find it useful. So, If you want to run * at bootup, and you have chan_h323, (a) then you should modyfy init.asterisk script with the path variables (shown below) and copy it to /etc/init.d, rename to asterisk (or anything) (b) Then do chkconfig --add asterisk and (c) chkconfig asterisk on (d) Now reboot and asterisk will start as a service Merry Christmas. Cheers SW -Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 24, 2003 2:25 PM To: SW Subject: Re: [Asterisk-Users] when * start at bootup chan_h323 fails to load You answered your own question here. Your startup environment does not use /etc/profile, so you have to copy those same directives into the asterisk startup script, so its environment is properly setup. Jeremy SW wrote: Hi Jeremy, I did read the README. Infact I knew you would love to RTFM :). Actually, I created the environment for BASH, exactly the way you asked to do. The question here is; chan_h323 get started, when I login as root and when I run the same script that I have in /etc/init.d. But it complaints when it is run at the boot time(so the path is good for user root, but not good when it is started at boot time). So, I must be doing something wrong in setting the environment, which seems only effective when logged in as root. I am running rh 9, and I put those path variables in /etc/profile. Here is my /etc/profile HOSTNAME=`/bin/hostname` HISTSIZE=1000 if [ -z $INPUTRC -a ! -f $HOME/.inputrc ]; then INPUTRC=/etc/inputrc fi export PATH USER LOGNAME MAIL HOSTNAME HISTSIZE INPUTRC for i in /etc/profile.d/*.sh ; do if [ -r $i ]; then . $i fi done unset i PWLIBDIR=/root/pwlib export PWLIBDIR OPENH323DIR=/root/openh323 export OPENH323DIR LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib export LD_LIBRARY_PATH also, echo $LD_LIBRARY_PATH gives me what it is. [EMAIL PROTECTED] sath]$ echo $LD_LIBRARY_PATH /root/pwlib/lib:/root/openh323/lib Is there any other log where we can take a closer look ? Would a complete clean and make of pwlib and openh323 would help? Things work fine, as far as call processing is concern, so I am reluctant to mess the installation again. Cheers SW Date: Wed, 24 Dec 2003 10:15:04 -0500 From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] when * start at bootup chan_h323 fails to load Reply-To: [EMAIL PROTECTED] SW wrote: (ast_load_resource): libpt_linux_x86_r.so.1.5.2: cannot open shared object file: No such file or directory Dec 23 23:33:52 WARNING[1074494176]: File loader.c, Line 407 (load_modules): Loading module chan_h323.so failed! RTFM cat /path/to/asterisk/channels/h323/README Jeremy McNamara libpt_linux_x86_r.so.1: cannot open shared object file: No such file or directory You have not set the LD_LIBRARY_PATH environment variable. Example environment for sh/bash: PWLIBDIR=$HOME/pwlib export PWLIBDIR OPENH323DIR=$HOME/openh323 export OPENH323DIR LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib export LD_LIBRARY_PATH We recomend puting the above directives into your /etc/profile so you do not have to remember to export those values every time you want to recompile. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Merry IAXmas
We wish you a merry IAXmas We wish you a merry IAXmas We wish you a merry IAXmas and a happy new year! From all of us in PK, Merry Xmas astmasters! May 2004 bring freedom from SIP/H323/MGCP/SCCP and all other junk protocols and may you realize the true spirit of IAX! - wasim and a special thank you to mr spencer and digium! you rock ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Quality Survey.... :P
On Wednesday 24 December 2003 20:14, Michael Welter wrote: Besides the ata186, which phone is next up the food chain? We are testing the Sipura SPA2000 and so far so good. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unlocking Vonage ATA 186
Vonage is running the latest 2.16-2 firmware. No longer applicable. cameron. On Wed, 24 Dec 2003, Doug Shubert wrote: this security hole has been around for some time http://www.securiteam.com/securitynews/5PP0G0K75U.html Lion Templin wrote: In the process of investigating a Cisco ATA 186 that was locked by Vonage, I found that you can still unlock the device yourself. But there's a catch. The device's design has a great plus: a DIP32 *socketed* SST28SF040A flash chip. I found an 8 digit unlock code at 0x03FA71-0x03FA78. I do not know if that is a standard location. If you have the equipment, you're in luck. But IMHO, the $15 fee is more than reasonable .. and certainly less than what it would cost to get a device to read/write these flash chips. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 1003 http://www.pulver.com/fwd/ ext. 83740 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
