Re: [Asterisk-Users] Zhone channel bank issues
Hi Michael, You might want to check the voltage settings on the FXS side of things. Also, are you using the correct signalling? (ground start, loop start, etc.) In the Zplex users guide, on page 41 you will see 2 sections on TTLP and RTLP. That might be of some help to you. Hey... You have caller ID working on that thing??? How did you do that? Let me know if you need a PDF copy of the manual -James On Mon, 10 Jan 2005 20:55:13 -0500, Michael Lyszczek [EMAIL PROTECTED] wrote: On Mon, 10 Jan 2005 12:51:49 -0500, Michael Lyszczek [EMAIL PROTECTED] wrote: Anyone have any issues like thisI am fwding broadvoice to zaptel,1 with my t100p and the t1 goes to a zhone zplex10b.. I can ring extension 1, which is pair 1 of the channel bank, but it doesnt recognize offhook and it keeps ringing the phone after I pick up. Also, its like each ring is like a seperate call as far as the callerid history goes. Anyone have any ideas? Michael Lyszczek ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Review of SIP Hard phones
I have had great experience so far with the Snom 190 and asterisk. I have used them at customer sites and we have them on our desks here at the office. I have tested the bugettone phone extensivly and I have had lots of problems with them. -James On Tue, 4 Jan 2005 17:15:54 -, Joao Pereira [EMAIL PROTECTED] wrote: check this link: http://www.iptel.org/info/products/sipphones.php João Pereira - Original Message - From: Michael Graves [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 04, 2005 5:00 PM Subject: Re: [Asterisk-Users] Review of SIP Hard phones On Tue, 4 Jan 2005 10:42:50 -0600, Michael B. Murdock wrote: Does anyone have a reference (link?) with reviews of the most popular SIP phones/ATA's can be found? We are looking to certify 3 or 4 VOIP phones and/or ATA devices for use with * and need to purchase one of each to test. If not, then what are the groups recommendations? The target customers are residential users and/or small business users for smaller telco's in the US. Currently on the short list are the following devices: Cisco ATA 18x series Sipura SPA-2000 Snom 190 Grandstream BudgeTone Any thoughts on the above as a starting point for testing? Are there any others that the group would recommend? What would be really helpful is if I could get a top 2 SIP phones recommendation and top 2 ATA's so that we don't waste a bunch of time looking at products that just don't cut it. I like www.atacomm.com. They've been very helpful both with hardware and Asterisk. I really like the Polycom phones which range from $200 - 400 USD. I also like the Zultys phones whch range from $100 - $350 USD. All are know to work well with *. While I have used the Sipura SPA-2000 3000. I have trended away from ATAs since SIP phones are more feature capable. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 8 pstn lines+ on Asterisk supported hardware.
Hi Hadi, I have been having troubles as well with the FXO/FXS cards from many installations at customers I have performed. My company has decided to forgo the FXO/FXS cards and now we use a T1 card with a channel bank. The versitility and expandibility is tremendous. Plus I enjoy the fact that I can punch down all the extensions into a 66 block. We have used the zhone 10B channel bank, but they are a pain to use. The best channel banks I have come across are from Rhino www.channelbanks.com . They are totally auto config, just plug it in and go. You might have to adjust the gain settings on them ,depending on the phones. They are comming out with a model that can do FXO along with FXS ports. -James On Mon, 3 Jan 2005 19:43:12 +0200, Hadi Jadallah [EMAIL PROTECTED] wrote: Hi all, I have this project that requires me to use 8 PSTN lines and possible more. I was thinking 2 TDM cards with FXO modules. The I got to read the Qs about FXO/FXS cards thread and that scared me. Can anybody recommend anything that is known to work ok with no mysterious problems? I was thinking OpenSwitch12 cards. What do you guys think? Any help is appreciated. Regards, Hadi -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.290 / Virus Database: 265.6.7 - Release Date: 12/30/2004 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 8 pstn lines+ on Asterisk supported hardware.
Some friendly FYI,I have to say that those are WAY overpriced. My company also imports those and I know off hand that the single port version costs $80. They are great boxes, made by welltech, in Taiwaan (spelling?) and are great ATA devices that work with Asterisk. We put them at customer sites sometimes. -James On Mon, 3 Jan 2005 16:19:08 -0800, Erik Espinoza [EMAIL PROTECTED] wrote: Or just get a couple of these: http://www.ipeya.com/VOIP_Products.htm (Specifically the 4 Ports FXO SIP VOIP-PSTN Gateway) Available from eBay at a discount at: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemcategory=61839item=5741966868rd=1 And do it all without worrying about irq's or the motherboard. Just let the device do it's job. Erik On Mon, 3 Jan 2005 15:48:47 -0500, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I have an Asterisk with 2 TDM with 8 FXO modules and I don't have any problems. One thing to look for is that the cards don't share any IRQ. Use a motherboard where you can assign IRQ to the PCI slot. I used an Intel board. Hope this help On Mon, 3 Jan 2005 19:43:12 +0200, Hadi Jadallah [EMAIL PROTECTED] wrote: Hi all, I have this project that requires me to use 8 PSTN lines and possible more. I was thinking 2 TDM cards with FXO modules. The I got to read the Qs about FXO/FXS cards thread and that scared me. Can anybody recommend anything that is known to work ok with no mysterious problems? I was thinking OpenSwitch12 cards. What do you guys think? Any help is appreciated. Regards, Hadi -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.290 / Virus Database: 265.6.7 - Release Date: 12/30/2004 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk dialing a Zap channel FXS instead of bridging to PSTN FXO
Hi All,
Channels 25-28 on a customers PBX are regular Zaptel FXO cards that
are hooked into 4 incomming phone lines. They are all in a group to do
automatic rollover for outgoing calls (if channel 25 is being used,
dial on channel 26, etc.).
Sometimes when a user is dialing a number, instead of bridging to one
of the FXO cards it goes and rings to Zap/1-1.
This doesnt occur all the time but some of the time, when it does
occur, I restart asterisk and it goes away for some time. I have also
tried changing the group number to something else, this doesnt seem to
help either.
I have a wait (w) before the numbers because the phone line doesnt
pick up right away and its to prevent asterisk from dialing before
there is a dial tone.
FYI, I have a rhino channel bank on the system going to a digium T100P
card, this is why my 4 FXO ports are so high.
Below I have snippets from my extensions.conf dial plan for the
outgoing context and my zapata.conf along with the error.
Error:
-- Executing Dial(Zap/6-1, Zap/g3/ww5632111) in new stack
-- Called g3/ww5632111
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
Extensions.conf context for outgoing calls
exten = _1NXXNXX,1,Dial(Zap/g3/ww${EXTEN})
exten = _NXXNXX,1,Dial(Zap/g3/w1${EXTEN})
exten = _NXX,1,Dial(Zap/g3/ww${EXTEN})
Zapata.conf snippet for the group
context=from-pstn
signalling=fxs_ks
callerid=asrecieved
;echocancel=yes
;echocancelwhenbridged=yes
;echotraining=400
rxgain=10.0
txgain=-4.5
group=3
channel = 25
context=from-pstn
signalling=fxs_ks
callerid=asrecieved
;echocancel=yes
;echocancelwhenbridged=yes
;echotraining=400
rxgain=12.0
txgain=-4.5
group=3
channel = 26
context=from-pstn
signalling=fxs_ks
callerid=asrecieved
;echocancel=yes
;echocancelwhenbridged=yes
;echotraining=400
rxgain=12.0
txgain=-4.5
group=3
channel = 27
context=from-pstn
signalling=fxs_ks
callerid=asrecieved
;echocancel=yes
;echocancelwhenbridged=yes
;echotraining=400
rxgain=12.0
txgain=-4.5
group=3
channel = 28
Thanks,
James
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[Asterisk-Users] Weird..bridging to Zap channel FXS instead of bridging to PSTN FXO on outgoing group
Hi All,
Channels 25-28 on a customers PBX are regular Zaptel FXO cards that
are hooked into 4 incomming phone lines. They are all in a group to do
automatic rollover for outgoing calls (if channel 25 is being used,
dial on channel 26, etc.).
Sometimes when a user is dialing a number, instead of bridging to one
of the FXO cards it goes and rings to Zap/1-1.
This doesnt occur all the time but some of the time, when it does
occur, I restart asterisk and it goes away for some time. I have also
tried changing the group number to something else, this doesnt seem to
help either.
I have a wait (w) before the numbers because the phone line doesnt
pick up right away and its to prevent asterisk from dialing before
there is a dial tone.
FYI, I have a rhino channel bank on the system going to a digium T100P
card, this is why my 4 FXO ports are so high.
Below I have snippets from my extensions.conf dial plan for the
outgoing context and my zapata.conf along with the error.
Error:
-- Executing Dial(Zap/6-1, Zap/g3/ww5632111) in new stack
-- Called g3/ww5632111
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
Extensions.conf context for outgoing calls
exten = _1NXXNXX,1,Dial(Zap/g3/ww${EXTEN})
exten = _NXXNXX,1,Dial(Zap/g3/w1${EXTEN})
exten = _NXX,1,Dial(Zap/g3/ww${EXTEN})
Zapata.conf snippet for the group
context=from-pstn
signalling=fxs_ks
callerid=asrecieved
;echocancel=yes
;echocancelwhenbridged=yes
;echotraining=400
rxgain=10.0
txgain=-4.5
group=3
channel = 25
context=from-pstn
signalling=fxs_ks
callerid=asrecieved
;echocancel=yes
;echocancelwhenbridged=yes
;echotraining=400
rxgain=12.0
txgain=-4.5
group=3
channel = 26
context=from-pstn
signalling=fxs_ks
callerid=asrecieved
;echocancel=yes
;echocancelwhenbridged=yes
;echotraining=400
rxgain=12.0
txgain=-4.5
group=3
channel = 27
context=from-pstn
signalling=fxs_ks
callerid=asrecieved
;echocancel=yes
;echocancelwhenbridged=yes
;echotraining=400
rxgain=12.0
txgain=-4.5
group=3
channel = 28
Thanks,
James
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RE: [Asterisk-Users] Can PSTN CallerID be fowarded to a SIP phone extension?
I am getting the following on my BT 100: t r This is when I make a call from another POTS phone to my PBX and then dial the SIP phones extension. From within the PBX I am able to recieve caller ID from SIP to SIP calls. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andrew Thompson Sent: Thursday, August 19, 2004 5:23 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Can PSTN CallerID be fowarded to a SIP phone extension? James Freire wrote: Sorry about that. I am in the US and using the Digium FXO TDM400 and I have enabled all the callerID options in my zapata.conf file. Have you enabled verbose debugging in the console and confirmed that you're receiving callerid from the PSTN? What are you getting on your SIP phones as callerid? - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can PSTN CallerID be fowarded to a SIP phone extension?
Sorry about this. I forgot to include the error from the CLI upon recieving an incomming call. Aug 20 08:37:53 NOTICE[622610]: chan_zap.c:5053 ss_thread: Got event 2 (Ring/Answered)... -- Detected ring pattern: 338,0,0 Aug 20 08:38:00 WARNING[622610]: chan_zap.c:5124 ss_thread: CallerID returned with error on channel 'Zap/8-1' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andrew Thompson Sent: Thursday, August 19, 2004 5:23 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Can PSTN CallerID be fowarded to a SIP phone extension? James Freire wrote: Sorry about that. I am in the US and using the Digium FXO TDM400 and I have enabled all the callerID options in my zapata.conf file. Have you enabled verbose debugging in the console and confirmed that you're receiving callerid from the PSTN? What are you getting on your SIP phones as callerid? - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk PBX Functions via SIP phone
Hi All, I am using a Grandstream BT100 and I have been trying to get the PBX features to work for DND, call foward, etc. These functions do work when I use my POTS phones hooked up to my Zap cards. But I cannot get the PBX functions (ie *78, *79) to work using my SIP phones. Is there a feature that has to be enabled to do this? I know these functions are available within the GS phone but all of them seem to just show the phone as being busy, even though, say, call foward is supposed to foward. It just makes the phone busy. I figure it would be easier just to have asterisk handling all those PBX functions. Thanks, James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk PBX Functions via SIP phone
Hi Craig, Thank you very much for the helpful information. I did enable that setting and it seems to have worked but not all the way. I do a *72 for an unconditional call forward + the number to forward to. Then when I dial the grandstream that has it enabled, asterisk just reponds that the extension is busy, the BT does not foward the call. I also get the following on the CLI -- Executing Dial(Zap/8-1, SIP/2000|20) in new stack -- Called 2000 -- Got SIP response 302 Moved Temporarily back from 64.201.13.50 -- SIP/2000-42e8 is busy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Craig Guy Sent: Friday, August 20, 2004 12:36 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone Hi James, This is a feature that needs to be enabled on both the phones and on Asterisk. So after enabling on your BT100 you need to add 'cancallforward=yes' to each extension in sip.conf you would like to add this feature to as in :- [9500] context=internal type=friend username=9500 host=dynamic callerid=9500 disallow=all allow=ulaw allow=alaw dtmfmode=info mailbox=9500 callgroup=1 pickupgroup=1 cancallforward=yes Craig - Original Message - From: James Freire [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, August 21, 2004 12:09 AM Subject: [Asterisk-Users] Asterisk PBX Functions via SIP phone Hi All, I am using a Grandstream BT100 and I have been trying to get the PBX features to work for DND, call foward, etc. These functions do work when I use my POTS phones hooked up to my Zap cards. But I cannot get the PBX functions (ie *78, *79) to work using my SIP phones. Is there a feature that has to be enabled to do this? I know these functions are available within the GS phone but all of them seem to just show the phone as being busy, even though, say, call foward is supposed to foward. It just makes the phone busy. I figure it would be easier just to have asterisk handling all those PBX functions. Thanks, James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk PBX Functions via SIP phone
I am suprised that one would have to create a dialplan since its an already built in
function that works with regular POTS phones. Or is it because of the way DTMF is sent
via SIP?
Someone correct me if I'm wrong but I believe you'll need the dialplan for
this one...
What I envision is doing something like this...
[verticalservice]
exten = *78,1,DbGet(${dnd}=features/dnd)
exten = *78,2,DbPut(features/dnd=1)
exten = *78,3,Playback(pbx-dndenabled)
exten = *78,4,Hangup()
exten = *78,102,GotoIf($[${dnd} = '0')]?103:104)
exteh = *78,103,DbPut(features/dnd=1)
exten = *78,104,Playback(pbx-dndenabled)
exten = *78,105,Hangup()
exten = *79 ... etc...
Wouldn't you need to track each extension? something like:
exten = *78,1,DbGet(${dnd}=dnd/${CALLERIDNUM})
exten = *78,2,DbPut(dnd/${CALLERIDNUM}=1)
exten = *78,3,Playback(pbx-dndenabled)
exten = *78,4,Hangup()
etc.?
The wiki has an exmple for call forwarding:
http://www.voip-info.org/wiki-Asterisk+call+forwarding
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[Asterisk-Users] Can PSTN CallerID be fowarded to a SIP phone extension?
Title: Can PSTN CallerID be fowarded to a SIP phone extension? Hi All, I have a server setup with an incomming PSTN line and a bunch of Grandstream BT100 phones. Is there a way for asterisk to foward an incomming callerID from the PSTN to the SIP phone that is setup as an extension? We have a Voice menu setup for incomming calls and I would like to recieve the caller ID of the calls we are recieving after the incomming caller reaches their final destination. Thanks! -James
RE: [Asterisk-Users] Can PSTN CallerID be fowarded to a SIP phone extension?
Sorry about that. I am in the US and using the Digium FXO TDM400 and I have enabled all the callerID options in my zapata.conf file. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Walt Reed Sent: Thursday, August 19, 2004 12:33 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can PSTN CallerID be fowarded to a SIP phone extension? On Thu, Aug 19, 2004 at 12:07:09PM -0400, James Freire said: I have a server setup with an incomming PSTN line and a bunch of Grandstream BT100 phones. Is there a way for asterisk to foward an incomming callerID from the PSTN to the SIP phone that is setup as an extension? We have a Voice menu setup for incomming calls and I would like to recieve the caller ID of the calls we are recieving after the incomming caller reaches their final destination. Um, * does this by default if callerID is supported by the FXO interface that you are using, and if * supports the callerID format in your country. Need info. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Does Granstream BT100 Conference Button Work?
Title: Does Granstream BT100 Conference Button Work? Hi All, I have tried searching everywhere but I cannot find a definitive answer as to if and how the conference button works on the BT100. Could anyone be kind enough to fill me in on some info on how to use the conferencing feature, as well as any configuration in asterisk thats needed, on this phone? Thank you, James
RE: [Asterisk-Users] Does Granstream BT100 Conference Button Work?
Title: Does Granstream BT100 Conference Button Work? Could I use the Flash button to do conferencing then??? If so.. how? -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Chris ShawSent: Thursday, August 19, 2004 4:28 PMTo: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Does Granstream BT100 Conference Button Work? Nope, it does nothing... It's not an * problem either, the button just does nothing... I think they're planning on making it work in a future release, don't quote me on that... for now it just occupies space.. -Chris - Original Message - From: James Freire To: [EMAIL PROTECTED] Sent: Thursday, August 19, 2004 12:53 PM Subject: [Asterisk-Users] Does Granstream BT100 Conference Button Work? Hi All, I have tried searching everywhere but I cannot find a definitive answer as to if and how the conference button works on the BT100. Could anyone be kind enough to fill me in on some info on how to use the conferencing feature, as well as any configuration in asterisk thats needed, on this phone? Thank you, James
[Asterisk-Users] Formatting in sip.conf...can you have 2 @ signs for register?
Title: Formatting in sip.conf...can you have 2 @ signs for register? Hi All, I am trying to setup another sip trunk in addition to what I am already using. The sip provider we are using right now gives you your username as your email address. So IE. If my email is [EMAIL PROTECTED] that is my username . Now... When I put this in the sip.conf file I have found that Asterisk is not able to parse it correctly and instantly goes to the email server to authenticate the sip user upon registration Here is the line below in my sip.conf file register = [EMAIL PROTECTED]:[EMAIL PROTECTED] THe error is below Aug 16 11:30:05 NOTICE[114695]: chan_sip.c:3922 sip_reg_timeout: Registration for '[EMAIL PROTECTED]@sip.voipamericas.com' timed out, trying again Aug 16 11:30:06 NOTICE[114695]: chan_sip.c:6575 handle_response: Failed to authenticate on REGISTER to 'sip:[EMAIL PROTECTED];tag=as1c528b93'
RE: [Asterisk-Users] Formatting in sip.conf...can you have 2 @ signs for register?
Hi Olle, I submitted the bug into the bugtracker. It is number 0002258 Thanks, James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Olle E. Johansson Sent: Monday, August 16, 2004 12:54 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Formatting in sip.conf...can you have 2 @ signs for register? James Freire wrote: Hi All, I am trying to setup another sip trunk in addition to what I am already using. The sip provider we are using right now gives you your username as your email address. So IE. If my email is [EMAIL PROTECTED] that is my username . Now... When I put this in the sip.conf file I have found that Asterisk is not able to parse it correctly and instantly goes to the email server to authenticate the sip user upon registration Here is the line below in my sip.conf file register = [EMAIL PROTECTED]:[EMAIL PROTECTED] THe error is below Aug 16 11:30:05 NOTICE[114695]: chan_sip.c:3922 sip_reg_timeout: Registration for '[EMAIL PROTECTED]@sip.voipamericas.com' timed out, trying again Aug 16 11:30:06 NOTICE[114695]: chan_sip.c:6575 handle_response: Failed to authenticate on REGISTER to 'sip:[EMAIL PROTECTED];tag=as1c528b93' That's obviously an error. Please add it to the bug tracker and we'll solve it. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can Incomming CallerID be fowarded to a SIP phone extension?
Title: Can Incomming CallerID be fowarded to a SIP phone extension? Hi All, I have a server setup with an incomming PSTN line and a bunch of Grandstream BT100 phones. Is there a way for asterisk to foward an incomming callerID to the SIP phone that is setup as an extension? We have a Voice menu setup for incomming calls and I would like to recieve the caller ID of the calls we are recieving after the incomming caller reaches their final destination. Thanks! -James
[Asterisk-Users] Lots of Echo with SIP - Asterisk - PSTN
Hi all. I am having this echo problem on my SIP phones when I am making a call from SIP to a PSTN line through asterisk. The echo goes away eventually after a few seconds when the call starts but it is very aparent during the start of the call. I do have echo cancellation turned on in asterisk as well as on the sip device. This is an ATA and not a SIP phone like a grandstream. I am running asterisk on a PII 350mhz machine and I am not sure if that has anyhing to do with the lack of performance in echo cancelation. I have 4 Digium FXO and 4 FXS ports in this machine. I know it does take a good ammount of CPU power to do echo cancellation. Later today I am getting a faster machine but I just wanted to see if there were any settings or tuning I can do in the compilation of asterisk to get it to perform better. Thanks a lot, James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Lots of Echo with SIP - Asterisk - PSTN
Actually, I had just made another identical install using a 1.2ghz celeron and there is no more echo! THe sound is great and there is no noticable delay between the sip phone and the POTS phone its calling to on the outside world. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chad Brown Sent: Friday, August 06, 2004 12:45 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Lots of Echo with SIP - Asterisk - PSTN Hi James, I had the same problem. Setting echotraning=yes helped me. Also, like many people I had to tweak the rxgain and txgain. For me these setting worked well. rxgain=0.0 txgain=-4.0 If I tried to increase the rxgain SIP echo got really bad. Although I could hear the PSTN caller much better. I settled for less echo. ;-) Setting the txgain=-4.0 eliminated echo that my PSTN caller was experiencing. However, it seems that everyone needs different settings dependant upon many factors... http://www.voip-info.org/wiki-Asterisk+echo+cancellation Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Freire Sent: Friday, August 06, 2004 4:12 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Lots of Echo with SIP - Asterisk - PSTN Hi all. I am having this echo problem on my SIP phones when I am making a call from SIP to a PSTN line through asterisk. The echo goes away eventually after a few seconds when the call starts but it is very aparent during the start of the call. I do have echo cancellation turned on in asterisk as well as on the sip device. This is an ATA and not a SIP phone like a grandstream. I am running asterisk on a PII 350mhz machine and I am not sure if that has anyhing to do with the lack of performance in echo cancelation. I have 4 Digium FXO and 4 FXS ports in this machine. I know it does take a good ammount of CPU power to do echo cancellation. Later today I am getting a faster machine but I just wanted to see if there were any settings or tuning I can do in the compilation of asterisk to get it to perform better. Thanks a lot, James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Trouble compiling asterisk-addons MySQL
Title: Trouble compiling asterisk-addons MySQL Hi All, I am having trouble compiling the mysql addon for asterisk. I had downloaded the most recent version from CVS and placed it in /usr/src/ and I get the following error below. [EMAIL PROTECTED]:/usr/src/asterisk-addons# make install ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c` cdr_addon_mysql.c:33:19: mysql.h: No such file or directory cdr_addon_mysql.c:34:20: errmsg.h: No such file or directory for x in ; do install -m 755 $x /usr/lib/asterisk/modules ; done BTW. I have asterisk running just fine. Thanks, James
RE: [Asterisk-Users] Best Linux for Asterisk
Hi Andy, I have had tremendous success running Asterisk on Slackware linux version 9.1. Its very quick to install and I had absolutely no problem compiling the source code for Asterisk or anything else so far. I have asterisk running on 2 servers right now that use Slackware. -James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Eric Kirkland Sent: Wednesday, July 28, 2004 9:14 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Best Linux for Asterisk Hi folks; Can anyone recommend the best Linux OS (versions, etc) to run Asterisk? I'd like to be able to run the Text To Speech apps and some of the extended functions of the software (no phone hardware needed, all Voice over IP stuff)... I'm currently running Asterisk on Mandrake Linux (vesion 10 I think?) but I'm having difficulty compiling the TTS stuff. I'm just wondering if there's a widely used version that pretty much works with everything...? Andy --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Trouble compiling asterisk-addons MySQL
Well. I have seemed to get a little farther with the problem. I added in a line in to the Makefile of CFLAGS+=-I/usr/local/mysql/include/mysql Now I get an error that has to do with mysqlclient below... I have also included my entire Makefile below the error. Thanks [EMAIL PROTECTED]:/usr/src/asterisk-addons# make install ./mkdep -fPIC -I../asterisk -I/usr/local/mysql/include/mysql -D_GNU_SOURCE -I/usr/local/mysql/include`ls *.c` cc -fPIC -I../asterisk -I/usr/local/mysql/include/mysql -D_GNU_SOURCE -I/usr/local/mysql/include -c -o cdr_addon_mysql.o cdr_addon_mysql.c cc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o -lmysqlclient -lz -L/usr/local/mysql/lib /usr/lib/gcc-lib/i486-slackware-linux/3.2.3/../../../../i486-slackware-linux/bin/ld: cannot find -lmysqlclient collect2: ld returned 1 exit status make: *** [cdr_addon_mysql.so] Error 1 # # Asterisk -- A telephony toolkit for Linux. # # Makefile for CDR backends (dynamically loaded) # # Copyright (C) 1999, Mark Spencer # # Mark Spencer [EMAIL PROTECTED] # # This program is free software, distributed under the terms of # the GNU General Public License # MODS= CFLAGS+=-fPIC CFLAGS+=-I../asterisk CFLAGS+=-I/usr/local/mysql/include/mysql CFLAGS+=-D_GNU_SOURCE INSTALL=install INSTALL_PREFIX= ASTLIBDIR=$(INSTALL_PREFIX)/usr/lib/asterisk MODULES_DIR=$(ASTLIBDIR)/modules # # MySQL stuff... Autoconf anyone?? # MODS+=$(shell if [ -d /usr/local/mysql/include/mysql ] || [ -d /usr/include/mysql ] || [ -d /usr /local/include/mysql ] || [ -d /opt/mysql/include ]; then echo cdr_addon_mysql.so; fi) CFLAGS+=$(shell if [ -d /usr/local/mysql/include/mysql ]; then echo -I/usr/local/mysql/include ; fi) CFLAGS+=$(shell if [ -d /usr/include/mysql ]; then echo -I/usr/include/mysql; fi) CFLAGS+=$(shell if [ -d /usr/local/include/mysql ]; then echo -I/usr/local/include/mysql; fi) CFLAGS+=$(shell if [ -d /opt/mysql/include/mysql ]; then echo -I/opt/mysql/include/mysql; fi) MLFLAGS= MLFLAGS+=$(shell if [ -d /usr/lib/mysql ]; then echo -L/usr/lib/mysql; fi) MLFLAGS+=$(shell if [ -d /usr/local/mysql/lib ]; then echo -L/usr/local/mysql/lib; fi) MLFLAGS+=$(shell if [ -d /usr/local/lib/mysql ]; then echo -L/usr/local/lib/mysql; fi) MLFLAGS+=$(shell if [ -d /opt/mysql/lib/mysql ]; then echo -L/opt/mysql/lib/mysql; fi) all: depend $(MODS) install: all for x in $(MODS); do $(INSTALL) -m 755 $$x $(MODULES_DIR) ; done clean: rm -f *.so *.o .depend %.so : %.o $(CC) -shared -Xlinker -x -o $@ $ ifneq ($(wildcard .depend),) include .depend endif cdr_addon_mysql.so: cdr_addon_mysql.o $(CC) -shared -Xlinker -x -o $@ $ -lmysqlclient -lz $(MLFLAGS) depend: .depend .depend: ./mkdep $(CFLAGS) `ls *.c` -Original Message- From: Oleg A. Arkhangelsky [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 28, 2004 9:48 AM To: [EMAIL PROTECTED]; James Freire Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Trouble compiling asterisk-addons MySQL Hello James, Wednesday, July 28, 2004, 5:23:16 PM, you wrote: JF [EMAIL PROTECTED]:/usr/src/asterisk-addons# make install JF ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c` JF cdr_addon_mysql.c:33:19: mysql.h: No such file or directory JF cdr_addon_mysql.c:34:20: errmsg.h: No such file or directory JF for x in ; do install -m 755 $x /usr/lib/asterisk/modules ; done You need to install libmysqlclient-devel (or alike) package with relevant header files. -- Best regards, Olegmailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Trouble compiling asterisk-addons MySQL
Yep.. I already have the headers and required files. Here is what I am getting now with my Make file also below it. [EMAIL PROTECTED]:/usr/src/asterisk-addons# make install ./mkdep -fPIC -I../asterisk -I/usr/local/mysql/include/mysql -D_GNU_SOURCE -I/usr/local/mysql/include`ls *.c` cc -fPIC -I../asterisk -I/usr/local/mysql/include/mysql -D_GNU_SOURCE -I/usr/local/mysql/include -c -o cdr_addon_mysql.o cdr_addon_mysql.c cc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o -lmysqlclient -lz -L/usr/local/mysql/lib /usr/lib/gcc-lib/i486-slackware-linux/3.2.3/../../../../i486-slackware-linux/bin/ld: cannot find -lmysqlclient collect2: ld returned 1 exit status make: *** [cdr_addon_mysql.so] Error 1 # # Asterisk -- A telephony toolkit for Linux. # # Makefile for CDR backends (dynamically loaded) # # Copyright (C) 1999, Mark Spencer # # Mark Spencer [EMAIL PROTECTED] # # This program is free software, distributed under the terms of # the GNU General Public License # MODS= CFLAGS+=-fPIC CFLAGS+=-I../asterisk CFLAGS+=-I/usr/local/mysql/include/mysql CFLAGS+=-D_GNU_SOURCE INSTALL=install INSTALL_PREFIX= ASTLIBDIR=$(INSTALL_PREFIX)/usr/lib/asterisk MODULES_DIR=$(ASTLIBDIR)/modules # # MySQL stuff... Autoconf anyone?? # MODS+=$(shell if [ -d /usr/local/mysql/include/mysql ] || [ -d /usr/include/mysql ] || [ -d /usr /local/include/mysql ] || [ -d /opt/mysql/include ]; then echo cdr_addon_mysql.so; fi) CFLAGS+=$(shell if [ -d /usr/local/mysql/include/mysql ]; then echo -I/usr/local/mysql/include ; fi) CFLAGS+=$(shell if [ -d /usr/include/mysql ]; then echo -I/usr/include/mysql; fi) CFLAGS+=$(shell if [ -d /usr/local/include/mysql ]; then echo -I/usr/local/include/mysql; fi) CFLAGS+=$(shell if [ -d /opt/mysql/include/mysql ]; then echo -I/opt/mysql/include/mysql; fi) MLFLAGS= MLFLAGS+=$(shell if [ -d /usr/lib/mysql ]; then echo -L/usr/lib/mysql; fi) MLFLAGS+=$(shell if [ -d /usr/local/mysql/lib ]; then echo -L/usr/local/mysql/lib; fi) MLFLAGS+=$(shell if [ -d /usr/local/lib/mysql ]; then echo -L/usr/local/lib/mysql; fi) MLFLAGS+=$(shell if [ -d /opt/mysql/lib/mysql ]; then echo -L/opt/mysql/lib/mysql; fi) all: depend $(MODS) install: all for x in $(MODS); do $(INSTALL) -m 755 $$x $(MODULES_DIR) ; done clean: rm -f *.so *.o .depend %.so : %.o $(CC) -shared -Xlinker -x -o $@ $ ifneq ($(wildcard .depend),) include .depend endif cdr_addon_mysql.so: cdr_addon_mysql.o $(CC) -shared -Xlinker -x -o $@ $ -lmysqlclient -lz $(MLFLAGS) depend: .depend .depend: ./mkdep $(CFLAGS) `ls *.c` -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Oleg A. Arkhangelsky Sent: Wednesday, July 28, 2004 9:48 AM To: [EMAIL PROTECTED]; James Freire Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Trouble compiling asterisk-addons MySQL Hello James, Wednesday, July 28, 2004, 5:23:16 PM, you wrote: JF [EMAIL PROTECTED]:/usr/src/asterisk-addons# make install JF ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c` JF cdr_addon_mysql.c:33:19: mysql.h: No such file or directory JF cdr_addon_mysql.c:34:20: errmsg.h: No such file or directory JF for x in ; do install -m 755 $x /usr/lib/asterisk/modules ; done You need to install libmysqlclient-devel (or alike) package with relevant header files. -- Best regards, Olegmailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Gui client
Hi, The version of astgui is 1.0.2. I am using PHP version 4.3.4-4 installed on a debian 3.0 system (testing) from apt-get. I do not have any GLOBAL_VARS set in my environment. What should it be? I am not very familiar with PHP. I had installed this on an existing system but made sure to install correctly all of the required packages that were listed in the instructions. I also have a problem, I dont know if it is related or not where when I first open the admin page I cannot get in with my username of gs102 and password of test. I verified that the username and password were in the database in the phones table. Thanks a lot! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of mattf Sent: Friday, July 16, 2004 9:55 PM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Asterisk Gui client Hello, What version of the astguiclient suite are you using? What version of PHP are you using? Do you have GLOBAL_VARS turned on or off? It's very strange that being a POST all of the variables seem to be showing up on the URL like a GET would. also it doesn't sem to be submitting to the admin.php script like it should be. Did you follow the SCRATCH_INSTALL instructions or are you mostly installing this on an existing system? MATT--- PS- I wrote the astguiclient suite :) -Original Message- From: James Freire [mailto:[EMAIL PROTECTED] Sent: Friday, July 16, 2004 5:13 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk Gui client I have installed the Asterisk gui client that is available off of sourceforge.net. I was curious if anybody here has used it and what experiences they have had with it. I am having a problem with it, I am able to use the admin page except when I try to submit information to the server to add phones I get an error, The requested URL /astguiclient/method=POST was not found on this server. The directory /astguiclient does exist and works because that is where the php files are located and running from. The URL for this command, so you can see what its submiting, is: http://172.16.200.80/astguiclient/method=POST?ADD=2extension=dialplan_numb er=voicemail_id=phone_ip=computer_ip=server_ip=login=pass=status=ACTI VEactive=Yphone_type=fullname=company=picture=submit=submit I am running Apache/1.3.29 with php installed also. My guess is that there is a bug somewhere in the php code but I do not know php well enough to troubleshoot it. Thanks a lot for any help, James Freire ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cant compile Zaptel at all
I have been trying to compile Zaptel 1.0-RC1 that I just downloaded via tarball on my debian 3.0 system running a 2.4.26 kernel. I have all the headers, libraries and sources installed for the kernel along with the latest versions of GCC. I dont know what else to do to trouble shoot this so I have included the entire output below. Thanks a lot! -James Freire linux1:/usr/src/zaptel-1.0-RC1# make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DSTANDALONE_ZAPATA -c zaptel.c In file included from /usr/src/linux/include/linux/kernel.h:13, from zaptel.c:42: /usr/src/linux/include/linux/types.h:21: error: parse error before dev_t /usr/src/linux/include/linux/types.h:21: warning: type defaults to `int' in declaration of `dev_t' /usr/src/linux/include/linux/types.h:21: warning: data definition has no type or storage class In file included from /usr/include/asm/math_emu.h:4, from /usr/include/asm/processor.h:11, from /usr/src/linux/include/linux/prefetch.h:13, from /usr/src/linux/include/linux/list.h:6, from /usr/src/linux/include/linux/module.h:12, from zaptel.c:44: /usr/include/asm/sigcontext.h:79: error: parse error before '*' token /usr/include/asm/sigcontext.h:82: error: parse error before '}' token In file included from /usr/include/asm/processor.h:11, from /usr/src/linux/include/linux/prefetch.h:13, from /usr/src/linux/include/linux/list.h:6, from /usr/src/linux/include/linux/module.h:12, from zaptel.c:44: /usr/include/asm/math_emu.h:6: error: parse error before '*' token /usr/include/asm/math_emu.h:7: error: parse error before '*' token In file included from /usr/src/linux/include/linux/prefetch.h:13, from /usr/src/linux/include/linux/list.h:6, from /usr/src/linux/include/linux/module.h:12, from zaptel.c:44: /usr/include/asm/processor.h:421: error: parse error before '*' token /usr/include/asm/processor.h:427: error: parse error before '}' token In file included from zaptel.c:44: /usr/src/linux/include/linux/module.h:21:34: linux/modversions.h: No such file or directory In file included from /usr/src/linux/include/linux/fs.h:19, from /usr/src/linux/include/linux/capability.h:17, from /usr/src/linux/include/linux/binfmts.h:5, from /usr/src/linux/include/linux/sched.h:9, from /usr/src/linux/include/linux/mm.h:4, from /usr/src/linux/include/linux/slab.h:14, from /usr/src/linux/include/linux/proc_fs.h:5, from zaptel.c:45: /usr/src/linux/include/linux/dcache.h: In function `d_drop': /usr/src/linux/include/linux/dcache.h:149: warning: implicit declaration of function `spin_lock' /usr/src/linux/include/linux/dcache.h:152: warning: implicit declaration of function `spin_unlock' In file included from /usr/src/linux/include/linux/capability.h:17, from /usr/src/linux/include/linux/binfmts.h:5, from /usr/src/linux/include/linux/sched.h:9, from /usr/src/linux/include/linux/mm.h:4, from /usr/src/linux/include/linux/slab.h:14, from /usr/src/linux/include/linux/proc_fs.h:5, from zaptel.c:45: /usr/src/linux/include/linux/fs.h: At top level: /usr/src/linux/include/linux/fs.h:426: error: parse error before dev_t /usr/src/linux/include/linux/fs.h:426: warning: no semicolon at end of struct or union /usr/src/linux/include/linux/fs.h:429: error: parse error before '}' token /usr/src/linux/include/linux/fs.h:435: error: parse error before dev_t /usr/src/linux/include/linux/fs.h:435: warning: no semicolon at end of struct or union /usr/src/linux/include/linux/fs.h:440: error: parse error before '}' token In file included from /usr/src/linux/include/linux/reiserfs_fs_sb.h:8, from /usr/src/linux/include/linux/fs.h:731, from /usr/src/linux/include/linux/capability.h:17, from /usr/src/linux/include/linux/binfmts.h:5, from /usr/src/linux/include/linux/sched.h:9, from /usr/src/linux/include/linux/mm.h:4, from /usr/src/linux/include/linux/slab.h:14, from /usr/src/linux/include/linux/proc_fs.h:5, from zaptel.c:45: /usr/src/linux/include/linux/tqueue.h: In function `queue_task': /usr/src/linux/include/linux/tqueue.h:107: warning: implicit declaration of function `typecheck' /usr/src/linux/include/linux/tqueue.h:107: error
[Asterisk-Users] Asterisk Gui client
I have installed the Asterisk gui client that is available off of sourceforge.net. I was curious if anybody here has used it and what experiences they have had with it. I am having a problem with it, I am able to use the admin page except when I try to submit information to the server to add phones I get an error, The requested URL /astguiclient/method=POST was not found on this server. The directory /astguiclient does exist and works because that is where the php files are located and running from. The URL for this command, so you can see what its submiting, is: http://172.16.200.80/astguiclient/method=POST?ADD=2extension=dialplan_number=voicemail_id=phone_ip=computer_ip=server_ip=login=pass=status=ACTIVEactive=Yphone_type=fullname=company=picture=submit=submit I am running Apache/1.3.29 with php installed also. My guess is that there is a bug somewhere in the php code but I do not know php well enough to troubleshoot it. Thanks a lot for any help, James Freire ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
