Re: [Asterisk-Users] Re: Compressing a dialplan
On Wed, 25 Aug 2004, Maron Kristófersson wrote: Hmm, that raises a lot of questions for the script... How many contexts do you have? Do they include each other. Is there any kind of rule around the extensions... etc. All the extensions will just run the same macro so all these are just the same. What I did is that I converted the Swedish national numbering plan (E.164) with an AWK script to an extension like file. The purpose is being able to use early dialling for national calls (no overlap dialling available). It contains the starting digits and number lengths, like this: 04623 0462400XXX 0462401XXX 0462402XXX 0462403XXX 0462404XXX 0462405XXX 0462406XXX 0462407XXX 0462408XXX 0462409XXX 046241 046242 046243 046244 0462450XX 0462451XX 0462452XX 0462453XX 0462454XX 0462455XX 0462456XX 0462457XX 0462458XX 0462459XX 046246XXX 046247XXX 046248XXX 046249XXX 04625 0462600XXX 0462601XXX 0462602XXX 0462603XXX 0462604XXX 0462605XXX 0462606XXX 0462607XXX 0462608XXX 0462609XXX 046261 046262 046263 046264 046265 046266 046267 046268 046269 It's obvious that the following four lines will match exactly the same numbers as all the lines above: 0462[35] 04624[0-4] 04624[5-9]XXX 04626X That is the conversation I would like a macro/script to do. What I thought about was if there are any kind of regexp compression programs or something like that. -- Regards, Tobias Jönsson, Lund SE___ 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] GrandStream HT-486 ATA as VoIP Gateway
1. Be sure to update to the newest firmware if you want to use the MAC-clone feature. 2. PCs running behind the built-in NAT router often have problems with corrupt files downloaded and socket errors. conclusion: i'm not happy with it klaus Miroslav Nachev wrote: Hi, Can I use HT-486 as VoIP Gateway together with Asterisk? Are there any success experiences? ___ 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] Voip phones headsets
On Wed, 2004-08-25 at 21:38 +0100, neil wrote: Hi all, I wonder if anyone could recommend a voip phone that supports headset working which works with * and advise me of a supplier of same. If any suppliers wish to respond please do with pricing for 60 phones shipped to the UK. We use an snom 200 for telephone English lessons with a normal PC style one sided headset and are more than happy with it. -- Dave Cotton [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] GrandStream HT-486 ATA as VoIP Gateway
On Wed, 25 Aug 2004, Miroslav Nachev wrote: Can I use HT-486 as VoIP Gateway together with Asterisk? Are there any success experiences? I have one and it works quite well. Its echo cancelling is too good :), resulting in a feel of half duplex audio, and the router thrughput is aweful (they are just 10 Mbps Ethernet ports and the thrughput is about 2 Mbps). There are some bugs that the grandstream people do not seem to care about, for example you cannot send DTMF tones when you receive a call - only when you are the caller. It also just provides American indication tones which could be quite confusing to europeans. -- Regards, Tobias Jönsson, Lund SE___ 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] Parallel T1 cable?
el Flynn wrote: Basically somehow I'd split the incoming T1 line into two parallel lines and connect them to two * servers. I found some hardware that allows you to do this, instead of hacking the T1 cable (which may lead to an impedance mismatch, from what I've been told). They're made by GL communications and I've put it up on the Wiki at http://www.voip-info.org/tiki-index.php?page=Failover+switches They basically make T1/E1 multiport repeaters. The single version provides ten identical outputs from a single T1/E1 input. The dual version provides four outputs for each of two inputs. So you could use this to connect one or two T1/E1 to multiple Asterisk servers (where one server would of course not be running Asterisk until the other server is dead) Just wanted this to get on the archives in case anyone else gets into the same situation as I did. Cheers, Flynn ___ 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] Voip phones headsets
I don't have pricing, but I'm using Cisco 7940G phones with Plantronics Supra headsets and they work perfectly - no amp required either. Same story with 7960G The good thing about the Cisco phones is that you have 3 options - handset, headset or speaker. Plenty of other phones require you to pick up the handset to answer a call with your headset and you can't switch between them without swapping cables. -Shaun - Original Message - From: neil [EMAIL PROTECTED] Date: Wed, 25 Aug 2004 21:38:44 +0100 Subject: [Asterisk-Users] Voip phones headsets To: [EMAIL PROTECTED] Hi all, I wonder if anyone could recommend a voip phone that supports headset working which works with * and advise me of a supplier of same. If any suppliers wish to respond please do with pricing for 60 phones shipped to the UK. Thanks in advance Neil ___ 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] Distinctive Ring Cadences
just store the cids of your high paying accs and give them vip treatment or a different did to call in =) On Thu, 26 Aug 2004 12:39:49 +1200, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On 25 Aug 2004 at 21:34, Nicolas Gudino wrote: On Wed, 2004-08-25 at 20:38, Chris Shaw wrote: Cool! I could see this being very useful, for example you could have an IVR that says something like Please set the priority of your call, 1 for urgent, 2 for normal or 3 for low then if 1, bellcore-r4, if 2 bellcore-r3, if 1 bellcore-r1! What for? People will always hit 1 g That's why you kinda need to make it an after call thing. LOL you could even use it in a queue... I.E. caller id starts with rating of 50 (max 100, min 0) After call press 1 for annoying, 2 for useful Then every time you press 1 their rating goes down...which could cause the queue priority to be higher...so if someone calls in with a rating of 25 and someone else with 75 you answer the 75 first! :-) Matt ___ 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] system reboot often?
Michael George wrote: Well, we only want 3 TDM400s: 4 FXO and 8 FXS. That will fit in nearly any desktop PC. That's not the scale that should require multiple boxes. But the question is where does the IRQ sharing instability creep in? I would think that *someone* out there would have a * box with 2-4 Digium cards in it that might be willing to share their experience. If the Digium cards can only be reliably run in a machine with only 1 or 2 of them, then I need to know so we can plan appropriately. As Rich alluded to, it's a bit of a lottery. I have two identical P4 2.4 boxes with Intel 845 chipsets running updated, stripped down Redhat 7.3 and custom compiled kernels containing nothing more than is required for asterisk in a headless, ssh access only situation. All onboard sound, USB etc. is disabled in the BIOS. One box has 3 x TDM400P - 2 x FXO and 8 x FXS (latest rev) all on individual IRQs. Until a couple of months ago it had 2 x X100P and 8 x TDM400 FXS and required driver reloads about every couple of months over a 1 year period. I replaced the 2 X100s with the TDM FXOs for a few reasons including a hoped for improvement in reliability. In the 2 months since, I have had to reload the drivers once - the logs showed 4 error mesages, Ouch, part reset, restoring reality for each of the ports on one of the FXS cards. The second box has a single TDM with 4 x FXO which is IAX2 trunked to the first. It originally had 2 x X100Ps and gave no problems at all for a year. In the 2 months since replacing with the single TDM, the drivers have needed reloading once, with no sign of any errors in logs. Apart from the above, I have been very happy the system and have had no echo problems. Regards, Richard ___ 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] Out Dial Problem
Dear All, I just setup the Asterisk with E100P which it's no problem in Dial In but I have problem when outdial. The connection method is like this : E1 PRI -SIGNAL-1- MaxLink (PBX) -SIGNAL-2- E100P - Asterisk -- SIP \- Analog PHone Now when I tried to dial out by SIP X-Lite on Windows, it shows me Connect, Trying, Dialing and then hangup. I've found the log as the following : *CLI Aug 26 15:54:17 DEBUG[-125376]: chan_sip.c:2332 sip_alloc: Allocating new SIP call for [EMAIL PROTECTED] Aug 26 15:54:17 DEBUG[-125376]: chan_sip.c:5200 check_user_full: Setting NAT on RTP to 0 Aug 26 15:54:17 DEBUG[-125376]: chan_sip.c:817 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 46613: Found Aug 26 15:54:17 DEBUG[-125376]: chan_sip.c:5200 check_user_full: Setting NAT on RTP to 0 Aug 26 15:54:17 DEBUG[-125376]: chan_sip.c:6991 handle_request: Check for res for 2000 Aug 26 15:54:17 DEBUG[-125376]: chan_sip.c:1633 update_user_counter: Call from user '2000' is 1 out of 0 Aug 26 15:54:17 DEBUG[-125376]: chan_sip.c:4423 build_route: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060 Aug 26 15:54:17 DEBUG[-1260983376]: pbx.c:1255 pbx_extension_helper: Launching 'ChanIsAvail' Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:6491 zt_request: Using channel 17 Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:733 ast_hangup: Hanging up channel 'Zap/17-1' Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:1902 zt_hangup: zt_hangup (Zap/17-1) Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:2417 zt_setoption: Set option AUDIO MODE, value: ON(1) on Zap/17-1 Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:1930 zt_hangup: Hangup: channel: 17 index = 0, normal = 38, callwait = -1, thirdcall = -1 Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:2329 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/17-1 Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:1151 update_conf: Updated conferencing on 17, with 0 conference users Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:2411 zt_setoption: Set option AUDIO MODE, value: OFF(0) on Zap/17-1 -- Hungup 'Zap/17-1' Aug 26 15:54:17 DEBUG[-1260983376]: pbx.c:1255 pbx_extension_helper: Launching 'Cut' Aug 26 15:54:17 DEBUG[-1260983376]: pbx.c:1255 pbx_extension_helper: Launching 'Dial' Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:6491 zt_request: Using channel 17 -- Called 17/008522112 Urgent handler Urgent handler Urgent handler Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1699 ast_set_read_format: Set channel Zap/17-1 to read format ALAW Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1666 ast_set_write_format: Set channel SIP/2000-e12c to write format ALAW Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1666 ast_set_write_format: Set channel Zap/17-1 to write format ALAW Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1699 ast_set_read_format: Set channel SIP/2000-e12c to read form at ALAW Aug 26 15:54:17 DEBUG[-1260983376]: rtp.c:1156 ast_rtp_write: Ooh, format changed from UNKN to ALAW Aug 26 15:54:17 DEBUG[-1248367696]: chan_zap.c:1179 zt_enable_ec: No echocancellation requested -- Zap/17-1 is ringing Urgent handler Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1395 ast_indicate: Driver for channel 'SIP/2000-e12c' does not support indication 3, emulating it Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1510 ast_prod: Prodding channel 'SIP/2000-e12c' Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1666 ast_set_write_format: Set channel SIP/2000-e12c to write format SLINR Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1666 ast_set_write_format: Set channel SIP/2000-e12c to write format ALAW Aug 26 15:54:17 DEBUG[-1248367696]: chan_zap.c:1179 zt_enable_ec: No echocancellation requested -- Zap/17-1 answered SIP/2000-e12c Urgent handler Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1699 ast_set_read_format: Set channel SIP/2000-e12c to read format ALAW Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1666 ast_set_write_format: Set channel Zap/17-1 to write format ALAW Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1666 ast_set_write_format: Set channel SIP/2000-e12c to write format ALAW Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1699 ast_set_read_format: Set channel Zap/17-1 to read format ALAW Aug 26 15:54:17 DEBUG[-1260983376]: chan_sip.c:1824 sip_answer: sip_answer (SIP/2000-e12c) Aug 26 15:54:17 DEBUG[-125376]: chan_sip.c:817 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 46614: Found Aug 26 15:54:17 DEBUG[-1260983376]: rtp.c:378 ast_rtcp_read: Got RTCP report of 84 bytes Aug 26 15:54:17 DEBUG[-1260983376]: rtp.c:378 ast_rtcp_read: Got RTCP report of 118 bytes -- Channel 0/17, span 1 got hangup Urgent handler Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:2559 ast_channel_bridge: Bridge stops because we're zombie or need a soft hangup: c0=SIP/2000-e12c, c1=Zap/17- 1, flags: No,No,No,Yes Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:2679
[Asterisk-Users] I have on isdn phone and one isdn-card (ast-pbx) attached (to my line)...
and i would happy if it be possible if the isdn-phone rings (number xyz) the ast-pbx rings too (number xyz is configured in capi.conf too). can anyone help me with that or is it not possible. If the isdn-phone rings the ast-pbx do not. Can it be that my phone answers the call without the call is taken? nico ___ 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] RE: SMP Performance
How do you setup your snoms? i have 2 problems: 1. call waiting indicator do not running (busy tone on 2. call). 2. How can i monitor a line like a callmanager with the led on the function-buttons. Can you help ? nicolas mattf wrote: We are currently running 4 asterisk servers in production all running SMP and performance is better under SMP than non-SMP. Right now we are averaging just under 20,000 calls (both in and out) a day on those 4 servers. As for the BEST VOIP phone, that is certainly up for debate. Here are my opinions: Cisco phones work well but are expensive Polycom phones are extremely similar to the Ciscos but are much cheaper. 3com phones are tricky to set up with Asterisk Snom phones are very good but take some getting used to I don't know of many people who have successfully set up Nortel VOIP phones on asterisk Avaya as always is expensive for what you get Pingtel's are pretty but there are current and future support and compatibility issues so I've heard Mitel VOIP phones work but do not offer enough features to justify the cost right now Grandstream phones are cheap(enough said) Sipura Analog adapters are very configurable and much cheaper than Cisco ATA Hope this helps. MATT--- -Original Message- From: Tim Jackson [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SMP Performance 25 should be the max ever. This machine used to be my testbed server. I may end up swapping it out later for a 1U IBM, but I just wanted to make sure that in the meantime it'd be able to handle what we are doing with it. We bought it refurbished for $600 about a year ago. I was just wondering about the SMP part, I've been told that it doesn't work well with SMP, and then I've been told it works fine. I just wanted a 2nd or 3rd opinion before I went ahead and implemented this. Another dumb question, I've gotten the idea that the best phones out there are the Cisco 7960s, any other good phones out there that are decently priced? Nortel? 3Com? -Tim -Original Message- From: mattf [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 8:43 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SMP Performance There is nothing wrong with running Asterisk on SMP. It runs quite well actually. I'm assuming you just have the Quad Xeon 450mhz sitting around because you can't buy them new anymore, so it probably isn't costing you anything to use it. In which case it isn't a waste. If you are paying more than $800 for it, save it and just buy a new P4 for less. A $200 machine may not be able to handle 25 concurrent conversations, and may have some used or sub-standard parts in it, so that may not be the best choice. You should be able to have upto 25 channels running on this machine no problem, How many maximum conversations do you forsee running concurrently at one time on this system? MATT--- -Original Message- From: Matt Schulte [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 9:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SMP Performance Meaning Asterisk won't/can't take advantage of the four CPU's? Or it's overkill for this scenario? -Original Message- From: joachim [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 12:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SMP Performance Send me the quad and i'll send you a 200$ pc to do this job. The quad is heavily overpowered. Joachim. At 22:00 24/08/2004, you wrote: content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary=_=_NextPart_001_01C48A15.130BF232 We're looking at implementing Asterisk in our department in the near future, we're looking at anywhere from 15-25 extensions. The machine we were looking at running this on was a Quad Xeon 450mhz (2MB L2 Cache) w/ 1GB of ram. I've heard bad things about running Asterisk on SMP machines? Would we be running into any performance issues with this machine? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827 office (936)414-6723 mobile ___ 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]
[Asterisk-Users] Codec
Good day all I want to know what the best codec is to use for asteris for VOIP We have two towns connected with a 64k line that's going to do VOIP with astersik.At the moment with the default installation the quality is bad and the bandwith is high. Is this even a codec problem Pleas help ALtus ___ 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] Re: 2x HFC ISDN Cards - SuSE 9.1 - Problems with making calls
Hey where you buyed the isdn-cards? i tryed to get some without success. nico Marco Czudej wrote: Hello everyone, I bought 2 HFC-ISDN Cards and want to run the first card in NT-Mode an the second one in TE-Mode. Everything looks ok under SuSE 9.1, but I can't dial out. I removed one card, for testing purposes and want to run this one card in TE-Mode. I only want to make a call with my Grandstream BT-101 over Asterisk via ISDN. When I try to make a call I get: - Executing Dial(SIP/11-3ef2, Zap/g1/00MY-NUMBER) in new stack Aug 25 18:11:00 NOTICE[1117453232]: app_dial.c:727 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time - zap show channels says: - Chan Extension Context Language MusicOnHold pseudodefault 1default 2default - ztcfg -vvv tells: - Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. - Only one point in zttool I don't really understand: - Current Alarms: No alarms. Sync Source:Internally clocked a IRQ Misses: 0 a Bipolar Viol: 0 a Tx/Rx Levels: 0/ 0 a Total/Conf/Act: 3/ 3/ 0 - Conf = configured or conflicted? I try a Loop but nothing happend. TxA, TxB etc. are empty, too. Can someone help me? - I really need some sample configs, too. Which linux distribution runs smoothest with Asterisk? Thanks! Marco Czudej ___ Gesendet von Yahoo! Mail - Jetzt mit 100MB Speicher kostenlos - Hier anmelden: http://mail.yahoo.de ___ 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] GSM to BRI ISDN Gateway
hello steve, I am interested to know how this is illegal, where is it written? Regards Nana Yaw On Wed, 25 Aug 2004 12:17:21 +0100, Steve Kennedy wrote On Wed, Aug 25, 2004 at 12:00:09PM +0100, Chris Lee wrote: Miroslav Nachev wrote: Hi, I am looking for GSM to BRI ISDN Gateway. Any help? I was also looking for such things nd came across these guys: http://www.2n.cz/export they have a product or two for GSM and here is the one I found most likely to work for me (two GSM sim cards providing two ISDN channels on a BRI line): http://www.2n.cz/uploads/2/PAGES/C379.HTML But I still have to get hold of one for testing, the local supplier is moving offices and as such can not help me out in the short term. Just beware GSM gateways are not legal in the UK if you're offering service to 3rd parties !!! i.e. you can connect one up to say a local PBX and connect your local GSM traffic through it, but not to anyone outside your organisation. Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ 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 -- Nana Yaw -- ___ 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] Grandstream Firmware
hi, I am looking to upgrade the firmware on my GS phone but the site doesn't have the IP adress of the TFTP server anymore or anywhere to download the firmware.. Does anyone know this information? What is the current stable firmware version? Later.. ___ 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] I have on isdn phone and one isdn-card (ast-pbx)attached (to my line)...
and i would happy if it be possible if the isdn-phone rings (number xyz) the ast-pbx rings too (number xyz is configured in capi.conf too). can anyone help me with that or is it not possible. If the isdn-phone rings the ast-pbx do not. Can it be that my phone answers the call without the call is taken? nico ___ 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] Grandstream Firmware
WipeOut wrote: hi, I am looking to upgrade the firmware on my GS phone but the site doesn't have the IP adress of the TFTP server anymore or anywhere to download the firmware.. Does anyone know this information? Can get it off the web: http://hellofone.com/downloads/ What is the current stable firmware version? 1.0.5.11 -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers I do not try to dance better than anyone else. I only try to dance better than myself. ___ 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] Grandstream Firmware
Duane wrote: WipeOut wrote: hi, I am looking to upgrade the firmware on my GS phone but the site doesn't have the IP adress of the TFTP server anymore or anywhere to download the firmware.. Does anyone know this information? Can get it off the web: http://hellofone.com/downloads/ What is the current stable firmware version? 1.0.5.11 Do the ATA's and the phones use the same firmware? ___ 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
R: [Asterisk-Users] Grandstream Firmware
What is the current stable firmware version? 1.0.5.11 Do the ATA's and the phones use the same firmware? Yes, it's the same firmware -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.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
Re: [Asterisk-Users] GSM to BRI ISDN Gateway
In most countries the legislation which licences the cellular operators to use certain spectrum for certain types of communication also controls who may provide publicly offered (and sometime privately offered) interworking services. This may also apply to wireline services. It is very country dependant (possibly even region dependant), so you need to check the detailed rules in your location. You might, for example, be OK doing things with SMS, but not with voice. Another level of checks you need to make is the details of your agreement with your service provider. Even if the licencing regulations say it is OK, your contract may not. Service then tends to be abruptly cut at the most awkward moments, and there is nothing you can do about it. In general telecoms licensing legislation tends to say you cannot do anything unless you are explicitly allowed to, so beware. :-) Regards, Steve Nana Yaw wrote: hello steve, I am interested to know how this is illegal, where is it written? Regards Nana Yaw On Wed, 25 Aug 2004 12:17:21 +0100, Steve Kennedy wrote On Wed, Aug 25, 2004 at 12:00:09PM +0100, Chris Lee wrote: Miroslav Nachev wrote: Hi, I am looking for GSM to BRI ISDN Gateway. Any help? I was also looking for such things nd came across these guys: http://www.2n.cz/export they have a product or two for GSM and here is the one I found most likely to work for me (two GSM sim cards providing two ISDN channels on a BRI line): http://www.2n.cz/uploads/2/PAGES/C379.HTML But I still have to get hold of one for testing, the local supplier is moving offices and as such can not help me out in the short term. Just beware GSM gateways are not legal in the UK if you're offering service to 3rd parties !!! i.e. you can connect one up to say a local PBX and connect your local GSM traffic through it, but not to anyone outside your organisation. Steve ___ 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] GSM to BRI ISDN Gateway
Hi, How did you find it in your locality? Eg, I am in London and have a T Mobile and an O2 mobile. I will check with them to see what they say first of all. Regards Leslie On Thu, 26 Aug 2004 17:07:08 +0800, Steve Underwood wrote In most countries the legislation which licences the cellular operators to use certain spectrum for certain types of communication also controls who may provide publicly offered (and sometime privately offered) interworking services. This may also apply to wireline services. It is very country dependant (possibly even region dependant), so you need to check the detailed rules in your location. You might, for example, be OK doing things with SMS, but not with voice. Another level of checks you need to make is the details of your agreement with your service provider. Even if the licencing regulations say it is OK, your contract may not. Service then tends to be abruptly cut at the most awkward moments, and there is nothing you can do about it. In general telecoms licensing legislation tends to say you cannot do anything unless you are explicitly allowed to, so beware. :-) Regards, Steve Nana Yaw wrote: hello steve, I am interested to know how this is illegal, where is it written? Regards Nana Yaw On Wed, 25 Aug 2004 12:17:21 +0100, Steve Kennedy wrote On Wed, Aug 25, 2004 at 12:00:09PM +0100, Chris Lee wrote: Miroslav Nachev wrote: Hi, I am looking for GSM to BRI ISDN Gateway. Any help? I was also looking for such things nd came across these guys: http://www.2n.cz/export they have a product or two for GSM and here is the one I found most likely to work for me (two GSM sim cards providing two ISDN channels on a BRI line): http://www.2n.cz/uploads/2/PAGES/C379.HTML But I still have to get hold of one for testing, the local supplier is moving offices and as such can not help me out in the short term. Just beware GSM gateways are not legal in the UK if you're offering service to 3rd parties !!! i.e. you can connect one up to say a local PBX and connect your local GSM traffic through it, but not to anyone outside your organisation. Steve ___ 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 -- Nana Yaw -- ___ 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] TDM400P + 1FXS + 1FXO for sale
Anyone interested in buying a TDM400P + 1FXS + 1FXO bundle drop me an email. I thought I'd offer here before I ebay it. -Dan ___ 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] for Lack of RTP activity in 0 seconds
Hi, we are using ser as registrar and proxy, * as gateway. Can someone explane me the * NOTICE Message chan_sip.c:7380 do_monitor: Disconnecting call 'SIP/sipgate.de-08352520' for lack of RTP activity in 0 seconds We got a lot of these messages and Call Hung ups right after the Notice. Greets Markus ___ 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] Faxing with SPANDSP or any other mean ? Is it possible ? Am I dreaming ?
Stopping in mid page is usually a timing problem. See the spandsp FAQ. Regards, Steve Jean-François Rousseau wrote: Hi , does anybody have successfully received a full fax with spandsp ? I keep having only about a quarter of the page and then the other part is garbage. Does anybody have any solution for this ? Right now I've tried: FAX --- IAXy --- ASTERISK --- SPANDSP And FAX --- PSTN --- X100P -- ASTERISK --- SPANDSP And both don't work, they give me only part of the page BTW, I also tried the fax on a local lan over an IAXy or on the PSTN with an X100P. Is there something I should know about faxing and theses two interfaces ? I also tried to Fax thru asterisk and it didn't work either FAX IAXy --- ASTERISK --- X100P --- PSTN --- FAX Finally my last test: FAX -- IAXy -- ASTERISK -- SIP (Iconnecthere) -- PSTN -- FAX didn't work too. Is there something I should know about faxing and Asterisk ? Should I use a Sipura SIP FXS ? P.S. I did start the ntp server to make sure timing was ok. Thanks in advance ___ Jean-François Rousseau Sys-Tech www.sys-tech.net [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] spandsp and certain (e.g. Canon) fax machines
Dear Steve the first line of t30.c is - #define LOG_FAX_AUDIO /* * SpanDSP - a series of DSP components for telephony * * t30.c - ITU T.30 FAX transfer processing * * Written by Steve Underwood [EMAIL PROTECTED] * * Copyright (C) 2003 Steve Underwood * -- I think is right uncomment but i dont see ant audio log under /tmp, do you think is possible no audio log? Thanks in advance Dimitri On Thursday 26 August 2004 06:15, Steve Underwood wrote: Hi, You are trying to receive from a Canon FAX machine. The problem I hope the change will fix is in sending *to* a Canon FAX machine. The user who found the bug in spandsp was trying to send to a Philips FAX machine. During negotiation the Philips sent a disconnect message, which is the same problem some people have with some Canon machines. It is not clear from your log why you have problems training the V.29 modem. Can you enable logging, by uncommenting the first line in t30.c, and send me the audio log files you will get in your /tmp directory. Regards, Steve reseaux wrote: Dear Steve i have try the SpanDSP (ver.k and latest Asterisk cvs) with the mod you have write below, but nothing my Canon Fax still dont send the fax: - -- Executing RxFAX(Zap/35-1, /home/user/testfax.tif) in new stack Changed from phase 0 to 1 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up TSI: 43 37 30 38 30 35 33 31 33 33 30 20 20 20 20 20 20 20 20 20 20 TSI without final frame tag Remote fax gave TSI as: 0331350807 DCS: 83 00 86 a0 00 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 10ms Get at 9600 Changed from phase 3 to 5 Fast carrier up Coarse carrier frequency 1728.81 (11) Fast carrier down Fast carrier up Coarse carrier frequency 1699.38 (86) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1700.06 (4917) Fast carrier down Fast carrier up Coarse carrier frequency 1699.33 (86) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier 1700.09 (4915) Fast carrier down Fast carrier up Coarse carrier frequency 1699.33 (86) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1700.07 (2924) Fast carrier down -- Executing NoOp(Zap/35-1, DIALEDTIME=) in new stack -- Executing NoOp(Zap/35-1, ANSWEREDTIME=) in new stack -- Executing Hangup(Zap/35-1, ) in new stack I dont know how to debug more, you can give more help to trace the problem? Thanks in advance. Dimitri On Wednesday 25 August 2004 11:34, Steve Underwood wrote: Hi, Several people have reported problems sending faxes from spandsp-0.0.1k to Canon FAX machines. A spandsp user had the same problem with another make of FAX machine, and traced the problem to a bug in the file t30.c of spandsp. Line 542 says s-t4.rx_file[0] where it should say s-t4.tx_file[0]. This fixes his problem, and I suspect it will also fix the Canon fax machine problem. Can someone having problems with Canon machines try this change, and tell me the result? Regards, Steve ___ 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 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] GSM to BRI ISDN Gateway
Nana Yaw wrote: Hi, How did you find it in your locality? Eg, I am in London and have a T Mobile and an O2 mobile. I will check with them to see what they say first of all. Regards Leslie Do you know a means to get a complete and honest answer from a telco? :-) Regards, Steve ___ 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] spandsp and certain (e.g. Canon) fax machines
Hi, If that is what you are running, you should be getting audio log files. The have names like /tmp/fax-rx-audio-date and /tmp/fax-tx-audio-date. I need a matching pair and the console log for investigation. Regards, Steve reseaux wrote: Dear Steve the first line of t30.c is - #define LOG_FAX_AUDIO /* * SpanDSP - a series of DSP components for telephony * * t30.c - ITU T.30 FAX transfer processing * * Written by Steve Underwood [EMAIL PROTECTED] * * Copyright (C) 2003 Steve Underwood * -- I think is right uncomment but i dont see ant audio log under /tmp, do you think is possible no audio log? Thanks in advance Dimitri On Thursday 26 August 2004 06:15, Steve Underwood wrote: Hi, You are trying to receive from a Canon FAX machine. The problem I hope the change will fix is in sending *to* a Canon FAX machine. The user who found the bug in spandsp was trying to send to a Philips FAX machine. During negotiation the Philips sent a disconnect message, which is the same problem some people have with some Canon machines. It is not clear from your log why you have problems training the V.29 modem. Can you enable logging, by uncommenting the first line in t30.c, and send me the audio log files you will get in your /tmp directory. Regards, Steve reseaux wrote: Dear Steve i have try the SpanDSP (ver.k and latest Asterisk cvs) with the mod you have write below, but nothing my Canon Fax still dont send the fax: - -- Executing RxFAX(Zap/35-1, /home/user/testfax.tif) in new stack Changed from phase 0 to 1 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up TSI: 43 37 30 38 30 35 33 31 33 33 30 20 20 20 20 20 20 20 20 20 20 TSI without final frame tag Remote fax gave TSI as: 0331350807 DCS: 83 00 86 a0 00 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 10ms Get at 9600 Changed from phase 3 to 5 Fast carrier up Coarse carrier frequency 1728.81 (11) Fast carrier down Fast carrier up Coarse carrier frequency 1699.38 (86) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1700.06 (4917) Fast carrier down Fast carrier up Coarse carrier frequency 1699.33 (86) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier 1700.09 (4915) Fast carrier down Fast carrier up Coarse carrier frequency 1699.33 (86) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1700.07 (2924) Fast carrier down -- Executing NoOp(Zap/35-1, DIALEDTIME=) in new stack -- Executing NoOp(Zap/35-1, ANSWEREDTIME=) in new stack -- Executing Hangup(Zap/35-1, ) in new stack I dont know how to debug more, you can give more help to trace the problem? Thanks in advance. Dimitri On Wednesday 25 August 2004 11:34, Steve Underwood wrote: Hi, Several people have reported problems sending faxes from spandsp-0.0.1k to Canon FAX machines. A spandsp user had the same problem with another make of FAX machine, and traced the problem to a bug in the file t30.c of spandsp. Line 542 says s-t4.rx_file[0] where it should say s-t4.tx_file[0]. This fixes his problem, and I suspect it will also fix the Canon fax machine problem. Can someone having problems with Canon machines try this change, and tell me the result? Regards, Steve ___ 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] spandsp and certain (e.g. Canon) fax machines
How to force * to write txfax console log into file ? On Thu, 2004-08-26 at 13:24, Steve Underwood wrote: Hi, If that is what you are running, you should be getting audio log files. The have names like /tmp/fax-rx-audio-date and /tmp/fax-tx-audio-date. I need a matching pair and the console log for investigation. Regards, Steve reseaux wrote: Dear Steve the first line of t30.c is - #define LOG_FAX_AUDIO /* * SpanDSP - a series of DSP components for telephony * * t30.c - ITU T.30 FAX transfer processing * * Written by Steve Underwood [EMAIL PROTECTED] * * Copyright (C) 2003 Steve Underwood * -- I think is right uncomment but i dont see ant audio log under /tmp, do you think is possible no audio log? Thanks in advance Dimitri On Thursday 26 August 2004 06:15, Steve Underwood wrote: Hi, You are trying to receive from a Canon FAX machine. The problem I hope the change will fix is in sending *to* a Canon FAX machine. The user who found the bug in spandsp was trying to send to a Philips FAX machine. During negotiation the Philips sent a disconnect message, which is the same problem some people have with some Canon machines. It is not clear from your log why you have problems training the V.29 modem. Can you enable logging, by uncommenting the first line in t30.c, and send me the audio log files you will get in your /tmp directory. Regards, Steve reseaux wrote: Dear Steve i have try the SpanDSP (ver.k and latest Asterisk cvs) with the mod you have write below, but nothing my Canon Fax still dont send the fax: - -- Executing RxFAX(Zap/35-1, /home/user/testfax.tif) in new stack Changed from phase 0 to 1 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up TSI: 43 37 30 38 30 35 33 31 33 33 30 20 20 20 20 20 20 20 20 20 20 TSI without final frame tag Remote fax gave TSI as: 0331350807 DCS: 83 00 86 a0 00 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 10ms Get at 9600 Changed from phase 3 to 5 Fast carrier up Coarse carrier frequency 1728.81 (11) Fast carrier down Fast carrier up Coarse carrier frequency 1699.38 (86) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1700.06 (4917) Fast carrier down Fast carrier up Coarse carrier frequency 1699.33 (86) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier 1700.09 (4915) Fast carrier down Fast carrier up Coarse carrier frequency 1699.33 (86) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1700.07 (2924) Fast carrier down -- Executing NoOp(Zap/35-1, DIALEDTIME=) in new stack -- Executing NoOp(Zap/35-1, ANSWEREDTIME=) in new stack -- Executing Hangup(Zap/35-1, ) in new stack I dont know how to debug more, you can give more help to trace the problem? Thanks in advance. Dimitri On Wednesday 25 August 2004 11:34, Steve Underwood wrote: Hi, Several people have reported problems sending faxes from spandsp-0.0.1k to Canon FAX machines. A spandsp user had the same problem with another make of FAX machine, and traced the problem to a bug in the file t30.c of spandsp. Line 542 says s-t4.rx_file[0] where it should say s-t4.tx_file[0]. This fixes his problem, and I suspect it will also fix the Canon fax machine problem. Can someone having problems with Canon machines try this change, and tell me the result? Regards, Steve ___ 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]
Re: [Asterisk-Users] spandsp and certain (e.g. Canon) fax machines
Dear Steve thanks for help, i now trace the log but no audio in /tmp - -- Starting simple switch on 'Zap/35-1' -- Executing RxFAX(Zap/35-1, /home/user/testfax.tif) in new stack Changed from phase 0 to 1 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 -- Remote UNIX connection -- Remote UNIX connection disconnected HDLC underflow in state 9 Changed from phase 4 to 3 T4 timeout in state 9 Changed from phase 3 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 T2 timeout Start receiving document Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 T4 timeout in state 9 Changed from phase 3 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 T2 timeout Start receiving document Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 T2 timeout Start receiving document Changed from phase 3 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 T4 timeout in state 9 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up TSI: 43 37 30 38 30 35 33 31 33 33 30 20 20 20 20 20 20 20 20 20 20 TSI without final frame tag Remote fax gave TSI as: 0331350807 DCS: 83 00 86 a0 00 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 10ms Get at 9600 Changed from phase 3 to 5 Fast carrier up Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe Floating point exception -- Thanks in adavance Dimitri On Thursday 26 August 2004 12:24, Steve Underwood wrote: Hi, If that is what you are running, you should be getting audio log files. The have names like /tmp/fax-rx-audio-date and /tmp/fax-tx-audio-date. I need a matching pair and the console log for investigation. Regards, Steve reseaux wrote: Dear
Re: [Asterisk-Users] GSM to BRI ISDN Gateway
On Thu, Aug 26, 2004 at 11:37:55AM +0300, Nana Yaw wrote: hello steve, I am interested to know how this is illegal, where is it written? In the UK there were various companies offering fixed to mobile gateways. The gateways generally were black boxes with PRI's on one side, and mobile connections on the other (GSM phones using retail SIMs). Apart from the obvious issues like CLI not being passed (user terminal equipment will only use the phone number associated with the SIM) these were pretty successful and lots of large telco's used them to pass traffic to the GSM networks. The situation came about because wholesale fixed to mobile pricing was so high (on average about 33p/min), while retails SIM deals could be done which would include various ammounts of traffic for a fixed fee. The black-boxes were clever in that they internally routed traffic to the various SIMs depending on time of day/etc rules so the inclusive minutes were always optimised. The mobile operators didn't like this and complained to Oftel the (at the time) UK Telecomms regulator, who found FOR the operators saying a mobile gateway was fixed therefore wasn't mobile and therefore broke the license that a mobile user operates under. However it only applied to people offering service to 3rd parties, as an individual/company could legitimately connect their mobile phone to their PC/PBX etc. The situation has changed now, since Ofcom (the new Super Regulator which regulates all aspect of communications including media) have forced the operators to considerably reduce their fixed to mobile rates. You can find the ruling on the Ofcom site (www.ofcom.org.uk). Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ 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] Noise on ZAP channel
I've been strugling with this for two days now and I'm making no progress at all. (I've checked been all over the WIKI and google). The issue is : I've installed a Digium X100P into my * box, configured it to call out over my BT analog line, every call I make is horribly noisy, there is a background hum on the line which sounds electrical. To troubleshoot I've plugged an analog line directly into the BT line there is no noise. I've * onto a brand spanking new machine incase it was the PSU in the PC causing the grief in the old machine, but still no luck. The PC is a P4 with 256MB of RAM and IDE drives. It's doing nothing except handle one SIP endpoint - which is a cisco 7960G. Codec is ulaw. Cat /proc/interpupts gives: CPU0 0: 155092 XT-PIC timer 1: 4 XT-PIC keyboard 2: 0 XT-PIC cascade 3: 0 XT-PIC ehci_hcd 5:1510338 XT-PIC wcfxo 7: 0 XT-PIC usb-uhci 8: 1 XT-PIC rtc 9: 14783 XT-PIC eth0 10: 0 XT-PIC usb-uhci 11: 0 XT-PIC usb-uhci 12: 7 XT-PIC PS/2 Mouse 14: 4597 XT-PIC ide0 15: 0 XT-PIC ide1 NMI: 0 ERR: 0 Here is the relevant part of /etc/zapatel.conf fxsls=1 loadzone=uk defaultzone=uk Here is the /etc/asterisk/zapatel.conf [channels] language=uk context=default signalling=fxs_ls hidecallerid=no cancallforward=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes callprogress=yes musiconhold=default channel=1 Any help would really be appriciated. Regards Matt ___ 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 media problem behind NAT
Hello All, I have a media problem while using sip communicator user agent with asterisk behind NAT.I had enabled the debug mode in asterisk and capture the results.I have attached the results with this mail.Can any one help me to fix the problem? Thanks in advance, Partha __ Do you Yahoo!? Yahoo! Mail is new and improved - Check it out! http://promotions.yahoo.com/new_mailSip read: REGISTER sip:asterisk ip:5060;transport=udp SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER From: 3002 sip:3002@asterisk ip:5060;transport=udp;tag=24957277 To: 3002 sip:3002@asterisk ip:5060;transport=udp Via: SIP/2.0/UDP 172.16.1.54:5060;branch=z9hG4bK850e4d5c6ff86d8844678ba62b0e89aa Max-Forwards: 70 Expires: 3600 Contact: 3002 sip:gateway1:5060;transport=udp Content-Length: 0 10 headers, 0 lines Using latest request as basis request Sending to 172.16.1.54 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.1.54:5060;branch=z9hG4bK850e4d5c6ff86d8844678ba62b0e89aa;received=gateway1 From: 3002 sip:3002@asterisk ip:5060;transport=udp;tag=24957277 To: 3002 sip:3002@asterisk ip:5060;transport=udp;tag=as4a5aa3e3 Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:3002@asterisk ip Content-Length: 0 to gateway1:5060 -- Registered SIP '3002' at gateway1 port 5060 expires 3600 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.1.54:5060;branch=z9hG4bK850e4d5c6ff86d8844678ba62b0e89aa;received=gateway1 From: 3002 sip:3002@asterisk ip:5060;transport=udp;tag=24957277 To: 3002 sip:3002@asterisk ip:5060;transport=udp;tag=as4a5aa3e3 Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: sip:3002@asterisk ip;expires=3600 Date: Thu, 26 Aug 2004 10:33:32 GMT Content-Length: 0 to gateway1:5060 Sip read: REGISTER sip:asterisk ip:5060;transport=udp SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER From: 3004 sip:3004@asterisk ip:5060;transport=udp;tag=13645178 To: 3004 sip:3004@asterisk ip:5060;transport=udp Via: SIP/2.0/UDP 192.168.1.38:5060;branch=z9hG4bK7ca01be9be9a2dca34277ce0ef3f5021 Max-Forwards: 70 Expires: 3600 Contact: 3004 sip:192.168.1.38:5060;transport=udp Content-Length: 0 10 headers, 0 lines Using latest request as basis request Sending to 192.168.1.38 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.38:5060;branch=z9hG4bK7ca01be9be9a2dca34277ce0ef3f5021;received=gateway2 ip From: 3004 sip:3004@asterisk ip:5060;transport=udp;tag=13645178 To: 3004 sip:3004@asterisk ip:5060;transport=udp;tag=as0934b948 Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:3004@asterisk ip Content-Length: 0 to gateway2 ip:5060 -- Registered SIP '3004' at gateway2 ip port 5060 expires 3600 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.38:5060;branch=z9hG4bK7ca01be9be9a2dca34277ce0ef3f5021;received=gateway2 ip From: 3004 sip:3004@asterisk ip:5060;transport=udp;tag=13645178 To: 3004 sip:3004@asterisk ip:5060;transport=udp;tag=as0934b948 Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: sip:3004@asterisk ip;expires=3600 Date: Thu, 26 Aug 2004 10:33:45 GMT Content-Length: 0 to gateway2 ip:5060 Sip read: INVITE sip:3004@asterisk ip SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE From: 3002 sip:3002@asterisk ip:5060;transport=udp;tag=24957277 To: sip:3004@asterisk ip Via: SIP/2.0/UDP 172.16.1.54:5060;branch=z9hG4bK3f113fc0c05ec1deece622bd0ed4a521 Max-Forwards: 70 Contact: 3002 sip:gateway1:5060;transport=udp Content-Type: application/sdp Content-Length: 148 v=0 o=par 0 0 IN IP4 gateway1 s=- c=IN IP4 gateway1 t=0 0 m=audio 4 RTP/AVP 0 3 4 5 6 8 15 18 m=video 2 RTP/AVP 26 34 31 10 headers, 7 lines Using latest request as basis request Sending to 172.16.1.54 : 5060 (non-NAT) Found audio format UNKN Found audio format UNKN Found audio format ULAW Found audio format UNKN Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format UNKN Found video format UNKN Found video format UNKN Found video format UNKN Capabilities: us - 786446, them - 303/851968, combined - 786446 Non-codec capabilities: us - 1, them - 0, combined - 0 Looking for 3004 in default list_route: hop: sip:gateway1:5060;transport=udp Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.1.54:5060;branch=z9hG4bK3f113fc0c05ec1deece622bd0ed4a521;received=gateway1 From: 3002 sip:3002@asterisk ip:5060;transport=udp;tag=24957277 To: sip:3004@asterisk ip;tag=as5bca4b71 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:3004@asterisk ip Content-Length: 0 to gateway1:5060 --
Re: [Asterisk-Users] Noise on ZAP channel
Hi Matt, You're using the wrong signalling type, should be fxs_ks HTH Chris -- Chris -- E Mail: [EMAIL PROTECTED] SIP: [EMAIL PROTECTED] IAXTEL: 17003366726 On Thu, 26 Aug 2004, Matt wrote: I've been strugling with this for two days now and I'm making no progress at all. (I've checked been all over the WIKI and google). The issue is : I've installed a Digium X100P into my * box, configured it to call out over my BT analog line, every call I make is horribly noisy, there is a background hum on the line which sounds electrical. To troubleshoot I've plugged an analog line directly into the BT line there is no noise. I've * onto a brand spanking new machine incase it was the PSU in the PC causing the grief in the old machine, but still no luck. The PC is a P4 with 256MB of RAM and IDE drives. It's doing nothing except handle one SIP endpoint - which is a cisco 7960G. Codec is ulaw. Cat /proc/interpupts gives: CPU0 0: 155092 XT-PIC timer 1: 4 XT-PIC keyboard 2: 0 XT-PIC cascade 3: 0 XT-PIC ehci_hcd 5:1510338 XT-PIC wcfxo 7: 0 XT-PIC usb-uhci 8: 1 XT-PIC rtc 9: 14783 XT-PIC eth0 10: 0 XT-PIC usb-uhci 11: 0 XT-PIC usb-uhci 12: 7 XT-PIC PS/2 Mouse 14: 4597 XT-PIC ide0 15: 0 XT-PIC ide1 NMI: 0 ERR: 0 Here is the relevant part of /etc/zapatel.conf fxsls=1 loadzone=uk defaultzone=uk Here is the /etc/asterisk/zapatel.conf [channels] language=uk context=default signalling=fxs_ls hidecallerid=no cancallforward=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes callprogress=yes musiconhold=default channel=1 Any help would really be appriciated. Regards Matt ___ 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] chan_capi module
Hello! I have tried to compile the capi module (http://www.junghanns.net/asterisk/downloads/chan_capi.0.3.5.tar.gz) on fedora2 today. --- MAKEFILE -- ..EXPORT_ALL_VARIABLES: INSTALL_PREFIX= #ASTERISK_HEADER_DIR=$(INSTALL_PREFIX)/usr/include ASTERISK_HEADER_DIR=/usr/include/asterisk #MODULES_DIR=$(INSTALL_PREFIX)/usr/lib/asterisk/modules MODULES_DIR=/usr/lib/asterisk/modules/ PROC=$(shell uname -m) --- MAKEFILE SNIP END-- Make Error: --- In file included from /usr/include/time.h:38, from /usr/include/pthread.h:21, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/lib/gcc-lib/i386-redhat-linux/3.3.3/include/stddef.h:213: error: syntax error before typedefIn file included from /usr/include/pthread.h:21, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/include/time.h:60: error: syntax error before typedef /usr/include/time.h:74: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/time.h:76: error: syntax error before typedef /usr/include/time.h:129: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/time.h:131: error: syntax error before struct /usr/include/time.h:178: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/time.h:181: error: syntax error before extern /usr/include/time.h:181: error: syntax error before __THROW /usr/include/time.h:184: error: syntax error before __THROW /usr/include/time.h:188: error: syntax error before __THROW /usr/include/time.h:191: error: syntax error before __THROW /usr/include/time.h:199: error: syntax error before __THROW /usr/include/time.h:226: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/time.h:229: error: syntax error before extern /usr/include/time.h:229: error: syntax error before __THROW /usr/include/time.h:233: error: syntax error before __THROW /usr/include/time.h:248: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/time.h:251: error: syntax error before extern /usr/include/time.h:251: error: syntax error before __THROW /usr/include/time.h:254: error: syntax error before __THROW /usr/include/time.h:272: error: syntax error before extern In file included from /usr/include/pthread.h:24, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/include/signal.h:31: error: syntax error before __BEGIN_DECLS In file included from /usr/include/signal.h:33, from /usr/include/pthread.h:24, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/include/bits/sigset.h:23: error: syntax error before typedef In file included from /usr/include/bits/pthreadtypes.h:23, from /usr/include/pthread.h:25, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/include/bits/sched.h:83: error: syntax error before struct In file included from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/include/pthread.h:59: error: syntax error before enum /usr/include/pthread.h:166: error: syntax error before __THROW /usr/include/pthread.h:169: error: syntax error before __THROW /usr/include/pthread.h:172: error: syntax error before __THROW /usr/include/pthread.h:186: error: syntax error before __THROW /usr/include/pthread.h:194: error: syntax error before __THROW /usr/include/pthread.h:197: error: syntax error before __THROW /usr/include/pthread.h:201: error: syntax error before __THROW /usr/include/pthread.h:205: error: syntax error before __THROW /usr/include/pthread.h:210: error: syntax error before __THROW /usr/include/pthread.h:216: error: syntax error before __THROW /usr/include/pthread.h:220: error: syntax error before __THROW /usr/include/pthread.h:225: error: syntax error before __THROW /usr/include/pthread.h:229: error: syntax error before __THROW /usr/include/pthread.h:234: error: syntax error before __THROW /usr/include/pthread.h:238: error: syntax error before __THROW /usr/include/pthread.h:242: error: syntax error before __THROW /usr/include/pthread.h:260: error: syntax error before __THROW /usr/include/pthread.h:265: error: syntax error before __THROW /usr/include/pthread.h:284: error: syntax error before __THROW /usr/include/pthread.h:289: error: syntax error before __THROW /usr/include/pthread.h:304: error: syntax error before __THROW /usr/include/pthread.h:310: error: syntax error before __THROW /usr/include/pthread.h:334: error: syntax error before __THROW /usr/include/pthread.h:337: error: syntax error before __THROW /usr/include/pthread.h:340: error: syntax error before __THROW /usr/include/pthread.h:343: error: syntax error before __THROW /usr/include/pthread.h:353: error: syntax error before __THROW /usr/include/pthread.h:360: error: syntax error before __THROW /usr/include/pthread.h:363: error: syntax error before __THROW
Re: [Asterisk-Users] spandsp and certain (e.g. Canon) fax machines
Hi, The following part of your log: Fast carrier up Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe Floating point exception would seem to indicate some problem with the libaudiofile on your machine. I guess libaudiofile must be installed, or you wouldn't even get that stuff in the log. Regards, Steve reseaux wrote: Dear Steve thanks for help, i now trace the log but no audio in /tmp - -- Starting simple switch on 'Zap/35-1' -- Executing RxFAX(Zap/35-1, /home/user/testfax.tif) in new stack Changed from phase 0 to 1 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 -- Remote UNIX connection -- Remote UNIX connection disconnected HDLC underflow in state 9 Changed from phase 4 to 3 T4 timeout in state 9 Changed from phase 3 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 T2 timeout Start receiving document Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 T4 timeout in state 9 Changed from phase 3 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 T2 timeout Start receiving document Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 T2 timeout Start receiving document Changed from phase 3 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 T4 timeout in state 9 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up TSI: 43 37 30 38 30 35 33 31 33 33 30 20 20 20 20 20 20 20 20 20 20 TSI without final frame tag Remote fax gave TSI as: 0331350807 DCS: 83 00 86 a0 00 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 10ms Get at 9600
Re: [Asterisk-Users] spandsp and certain (e.g. Canon) fax machines
Dear Steve i use Mandrake 9.2 with libaudiofile0-devel-0.2.3-6mdk libaudiofile0-0.2.3-6mdk do you think is correct version? After that message Asterisk crash.. Thanks Dimitri On Thursday 26 August 2004 13:33, Steve Underwood wrote: Hi, The following part of your log: Fast carrier up Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe Floating point exception would seem to indicate some problem with the libaudiofile on your machine. I guess libaudiofile must be installed, or you wouldn't even get that stuff in the log. Regards, Steve reseaux wrote: Dear Steve thanks for help, i now trace the log but no audio in /tmp - -- Starting simple switch on 'Zap/35-1' -- Executing RxFAX(Zap/35-1, /home/user/testfax.tif) in new stack Changed from phase 0 to 1 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 -- Remote UNIX connection -- Remote UNIX connection disconnected HDLC underflow in state 9 Changed from phase 4 to 3 T4 timeout in state 9 Changed from phase 3 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 T2 timeout Start receiving document Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 T4 timeout in state 9 Changed from phase 3 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 T2 timeout Start receiving document Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 T2 timeout Start receiving document Changed from phase 3 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 T4 timeout in state 9 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up TSI: 43 37 30 38 30 35 33 31 33 33 30 20 20 20 20 20 20 20 20 20 20 TSI without
RE: [Asterisk-Users] Noise on ZAP channel
Chris, Thanks, that's better, but there is still a faint buzzing in the background. Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Glover Sent: 26 August 2004 12:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Noise on ZAP channel Hi Matt, You're using the wrong signalling type, should be fxs_ks HTH Chris -- Chris -- E Mail: [EMAIL PROTECTED] SIP: [EMAIL PROTECTED] IAXTEL: 17003366726 On Thu, 26 Aug 2004, Matt wrote: I've been strugling with this for two days now and I'm making no progress at all. (I've checked been all over the WIKI and google). The issue is : I've installed a Digium X100P into my * box, configured it to call out over my BT analog line, every call I make is horribly noisy, there is a background hum on the line which sounds electrical. To troubleshoot I've plugged an analog line directly into the BT line there is no noise. I've * onto a brand spanking new machine incase it was the PSU in the PC causing the grief in the old machine, but still no luck. The PC is a P4 with 256MB of RAM and IDE drives. It's doing nothing except handle one SIP endpoint - which is a cisco 7960G. Codec is ulaw. Cat /proc/interpupts gives: CPU0 0: 155092 XT-PIC timer 1: 4 XT-PIC keyboard 2: 0 XT-PIC cascade 3: 0 XT-PIC ehci_hcd 5:1510338 XT-PIC wcfxo 7: 0 XT-PIC usb-uhci 8: 1 XT-PIC rtc 9: 14783 XT-PIC eth0 10: 0 XT-PIC usb-uhci 11: 0 XT-PIC usb-uhci 12: 7 XT-PIC PS/2 Mouse 14: 4597 XT-PIC ide0 15: 0 XT-PIC ide1 NMI: 0 ERR: 0 Here is the relevant part of /etc/zapatel.conf fxsls=1 loadzone=uk defaultzone=uk Here is the /etc/asterisk/zapatel.conf [channels] language=uk context=default signalling=fxs_ls hidecallerid=no cancallforward=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes callprogress=yes musiconhold=default channel=1 Any help would really be appriciated. Regards Matt ___ 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] Overhead Paging
All, I am currently implementing a VoIP PBX, and need to deal with the paging situation. I would prefer to do paging via overhead speakers. My plan is to connect a Paging Unit to an FXS port of an IAD, and assign an extension to that port. I would then simply be able to call that extension, and have my call patched through to the overhead speakers. Has anyone implemented this type of setup, if so, what type of paging unit did you deploy, did you require an external amplifier or power supply, and how many speakers were you able to connect to the unit? As it stands, I will need between 4 and 8 speakers, and some of the speakers will be 400 feet from the main telco closet. Any thoughts, comments, and suggestions that you can shed on this topic would be much appreciated. If you have other methods of implementing overhead paging, I would also be interested. -Brian ___ 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] system reboot often?
I have two identical P4 2.4 boxes with Intel 845 chipsets running updated, stripped down Redhat 7.3 and custom compiled kernels containing nothing more than is required for asterisk in a headless, ssh access only situation. All onboard sound, USB etc. is disabled in the BIOS. Would you mind maybe expanding upon the hardware configuration you are using and why? I, and I'm sure others, are curious as to what you are using. I haven't had to roll out any systems yet that require multiple Digium cards, but I'm sure the information would be quite useful as I've seen few posts regarding this issue. Thanks, Leif Madsen. http://www.asteriskdocs.org ___ 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] chan_capi module
i think your problem is above the pasted error message. i compiled chan_capi on fedora 2 just yesterday. only problem was that some isdn4k devel pkg was missing. install it and it'll probably work fine. On Thu, 26 Aug 2004 13:32:06 +0200 (CEST), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello! I have tried to compile the capi module (http://www.junghanns.net/asterisk/downloads/chan_capi.0.3.5.tar.gz) on fedora2 today. --- MAKEFILE -- ...EXPORT_ALL_VARIABLES: INSTALL_PREFIX= #ASTERISK_HEADER_DIR=$(INSTALL_PREFIX)/usr/include ASTERISK_HEADER_DIR=/usr/include/asterisk #MODULES_DIR=$(INSTALL_PREFIX)/usr/lib/asterisk/modules MODULES_DIR=/usr/lib/asterisk/modules/ PROC=$(shell uname -m) --- MAKEFILE SNIP END-- Make Error: --- In file included from /usr/include/time.h:38, from /usr/include/pthread.h:21, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/lib/gcc-lib/i386-redhat-linux/3.3.3/include/stddef.h:213: error: syntax error before typedefIn file included from /usr/include/pthread.h:21, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/include/time.h:60: error: syntax error before typedef /usr/include/time.h:74: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/time.h:76: error: syntax error before typedef /usr/include/time.h:129: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/time.h:131: error: syntax error before struct /usr/include/time.h:178: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/time.h:181: error: syntax error before extern /usr/include/time.h:181: error: syntax error before __THROW /usr/include/time.h:184: error: syntax error before __THROW /usr/include/time.h:188: error: syntax error before __THROW /usr/include/time.h:191: error: syntax error before __THROW /usr/include/time.h:199: error: syntax error before __THROW /usr/include/time.h:226: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/time.h:229: error: syntax error before extern /usr/include/time.h:229: error: syntax error before __THROW /usr/include/time.h:233: error: syntax error before __THROW /usr/include/time.h:248: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/time.h:251: error: syntax error before extern /usr/include/time.h:251: error: syntax error before __THROW /usr/include/time.h:254: error: syntax error before __THROW /usr/include/time.h:272: error: syntax error before extern In file included from /usr/include/pthread.h:24, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/include/signal.h:31: error: syntax error before __BEGIN_DECLS In file included from /usr/include/signal.h:33, from /usr/include/pthread.h:24, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/include/bits/sigset.h:23: error: syntax error before typedef In file included from /usr/include/bits/pthreadtypes.h:23, from /usr/include/pthread.h:25, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/include/bits/sched.h:83: error: syntax error before struct In file included from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/include/pthread.h:59: error: syntax error before enum /usr/include/pthread.h:166: error: syntax error before __THROW /usr/include/pthread.h:169: error: syntax error before __THROW /usr/include/pthread.h:172: error: syntax error before __THROW /usr/include/pthread.h:186: error: syntax error before __THROW /usr/include/pthread.h:194: error: syntax error before __THROW /usr/include/pthread.h:197: error: syntax error before __THROW /usr/include/pthread.h:201: error: syntax error before __THROW /usr/include/pthread.h:205: error: syntax error before __THROW /usr/include/pthread.h:210: error: syntax error before __THROW /usr/include/pthread.h:216: error: syntax error before __THROW /usr/include/pthread.h:220: error: syntax error before __THROW /usr/include/pthread.h:225: error: syntax error before __THROW /usr/include/pthread.h:229: error: syntax error before __THROW /usr/include/pthread.h:234: error: syntax error before __THROW /usr/include/pthread.h:238: error: syntax error before __THROW /usr/include/pthread.h:242: error: syntax error before __THROW /usr/include/pthread.h:260: error: syntax error before __THROW /usr/include/pthread.h:265: error: syntax error before __THROW /usr/include/pthread.h:284: error: syntax error before __THROW /usr/include/pthread.h:289: error: syntax error before __THROW /usr/include/pthread.h:304: error: syntax error before __THROW /usr/include/pthread.h:310: error: syntax error before __THROW /usr/include/pthread.h:334: error: syntax error before __THROW
[Asterisk-Users] Re: Noise on ZAP channel
In article [EMAIL PROTECTED], Matt [EMAIL PROTECTED] wrote: Chris, Thanks, that's better, but there is still a faint buzzing in the background. [X100P on BT line] The X100P is poorly impedance-matched to UK phone lines, so will never work very well. The impedance matching is fixed in the hardware, so can't be configured by the software. The FXO modules for the TDM400P (e.g. TDM01B) have software-configurable impedance setting and I believe the zaptel code includes the settings for UK lines. So they should work much better, but I haven't tried one. Unfortunately, Telappliant don't seem to stock those yet, so you would have to order from the US. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] chan_capi module
i think your problem is above the pasted error message. i compiled chan_capi on fedora 2 just yesterday. only problem was that some isdn4k devel pkg was missing. install it and it'll probably work fine. I have just installed them: rpm -Uvh fedora-core2-full/Fedora/RPMS/isdn4k* and ran ldconfig afterwards. Still the same error. Which other libs or devel package could be missing? On Thu, 26 Aug 2004 13:32:06 +0200 (CEST), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello! I have tried to compile the capi module (http://www.junghanns.net/asterisk/downloads/chan_capi.0.3.5.tar.gz) on fedora2 today. --- MAKEFILE -- ...EXPORT_ALL_VARIABLES: INSTALL_PREFIX= #ASTERISK_HEADER_DIR=$(INSTALL_PREFIX)/usr/include ASTERISK_HEADER_DIR=/usr/include/asterisk #MODULES_DIR=$(INSTALL_PREFIX)/usr/lib/asterisk/modules MODULES_DIR=/usr/lib/asterisk/modules/ PROC=$(shell uname -m) --- MAKEFILE SNIP END-- Make Error: --- In file included from /usr/include/time.h:38, from /usr/include/pthread.h:21, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/lib/gcc-lib/i386-redhat-linux/3.3.3/include/stddef.h:213: error: syntax error before typedefIn file included from /usr/include/pthread.h:21, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/include/time.h:60: error: syntax error before typedef /usr/include/time.h:74: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/time.h:76: error: syntax error before typedef /usr/include/time.h:129: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/time.h:131: error: syntax error before struct /usr/include/time.h:178: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/time.h:181: error: syntax error before extern /usr/include/time.h:181: error: syntax error before __THROW /usr/include/time.h:184: error: syntax error before __THROW /usr/include/time.h:188: error: syntax error before __THROW /usr/include/time.h:191: error: syntax error before __THROW /usr/include/time.h:199: error: syntax error before __THROW /usr/include/time.h:226: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/time.h:229: error: syntax error before extern /usr/include/time.h:229: error: syntax error before __THROW /usr/include/time.h:233: error: syntax error before __THROW /usr/include/time.h:248: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/time.h:251: error: syntax error before extern /usr/include/time.h:251: error: syntax error before __THROW /usr/include/time.h:254: error: syntax error before __THROW /usr/include/time.h:272: error: syntax error before extern In file included from /usr/include/pthread.h:24, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/include/signal.h:31: error: syntax error before __BEGIN_DECLS In file included from /usr/include/signal.h:33, from /usr/include/pthread.h:24, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/include/bits/sigset.h:23: error: syntax error before typedef In file included from /usr/include/bits/pthreadtypes.h:23, from /usr/include/pthread.h:25, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/include/bits/sched.h:83: error: syntax error before struct In file included from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/include/pthread.h:59: error: syntax error before enum /usr/include/pthread.h:166: error: syntax error before __THROW /usr/include/pthread.h:169: error: syntax error before __THROW /usr/include/pthread.h:172: error: syntax error before __THROW /usr/include/pthread.h:186: error: syntax error before __THROW /usr/include/pthread.h:194: error: syntax error before __THROW /usr/include/pthread.h:197: error: syntax error before __THROW /usr/include/pthread.h:201: error: syntax error before __THROW /usr/include/pthread.h:205: error: syntax error before __THROW /usr/include/pthread.h:210: error: syntax error before __THROW /usr/include/pthread.h:216: error: syntax error before __THROW /usr/include/pthread.h:220: error: syntax error before __THROW /usr/include/pthread.h:225: error: syntax error before __THROW /usr/include/pthread.h:229: error: syntax error before __THROW /usr/include/pthread.h:234: error: syntax error before __THROW /usr/include/pthread.h:238: error: syntax error before __THROW /usr/include/pthread.h:242: error: syntax error before __THROW /usr/include/pthread.h:260: error: syntax error before __THROW /usr/include/pthread.h:265: error: syntax error before __THROW /usr/include/pthread.h:284: error: syntax error before __THROW /usr/include/pthread.h:289: error: syntax error before __THROW
RE: [Asterisk-Users] need help with zaptel.
Edward, I had the same problem I am running fedora core 2 with a 2.6.8-1.521 kernel at it is working now perfect. Craig wrote the following : I have had success with this using both the X100p (wcfxs and wcfxo) and TE410p (wct4xxp) under Redhat FC2 2.6.5. The instructions are on the wiki, do the following: ln -s /lib/modules/2.6.5-1.358/build linux-2.6 cd zaptel make clean make linux26 make install I made my softlink to /usr/src/linux-2.6.8-1.521 after that the compiling of Zaptel is working perfectly but you get the inserting module error. After that I changed the softlink to /lib/modules/2.8.9-1.521/build and now everything is working perfectly. But dont forget to recompile zaptel after changing the softlink Han From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Edward Huitt Sent: Saturday, August 21, 2004 9:22 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] need help with zaptel. Ok I am past the compile. Now when I try to modprobe I get FATAL: Error inserting zaptel (/lib/modules/2.6.5-1.358/misc/zaptel.ko): Invalid module format -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Blakely Sent: Saturday, August 21, 2004 7:00 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] need help with zaptel. Look in /usr/src. You should see a directory something similar to linux-2.6.1-1[as an example]. If you DON'T have a directory (or link to a directory) named linux-2.6, you should create one using the 'ln' command. In the case mentioned above, the command would be: ln -s /usr/src/linux-2.6.1-1 /usr/src/linux-2.6 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Edward Huitt Sent: Saturday, August 21, 2004 5:42 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] need help with zaptel. I cant get zaptel to make. I get this error: make: *** [linux26] Error 1. The previous line is: Link /usr/src/linux-2.6 to your kernel sources first! I am running Fedora Core2 Asterisk compiles fine. I am using my SIP phones. I would like to get my TDM400p working. ___ 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] GRSecurity and ALSA on a Gentoo Server
I've been working with Asterisk for about 2 months now and am doing well. However I decided to switch platforms from Fedora Core 1, that my predacessor was using, to Gentoo, for obvious reasons. It just seems faster and less bloated everything I need, nothing I don't. Anyways, I've read what the Wiki had to say about it and I was only confused on one thing, putting ALSA in my USE statement. It's a 1U server with no Sound Card. I did not choose to put ALSA in my USE flags as I don't have a sound card. But will Asterisk suffer in any way? I know that Asterisk is fully capable of running on a machine with No Sound card, my Fedora servers have no sound card, but by ommitting alsa in my USE flags, will Asterisk be compiled in a way that would make it less functional? My last question, sorry guys (and girls), is about the grsecurity in the 2.4 kernel (I chose 2.4 instead of 2.6). I set it to low for now, as it said it wouldn't cause any compatibility issues with 99% of the programs. Has anybody tried medium, or even high, with Asterisk? How secure can you get the kernel without interfering with Asterisk. This is just more of a comment, but if anybody see's anything wrong with it I'd like to know. I don't want to use the 0.9.0 ebuild (but I emerged it just to get the dependencies taken care of) so I emerge'd the CVS program so that I can upgrade libpri/zaptel/asterisk from 0.9.0 to the latest. The The Wiki mentions something about CVS and points to: http://bugs.gentoo.org/show_bug.cgi?id=33345 but that link is dead. I figured I'd just CVS Asterisk the normal way, do the make install and it should upgrade it. Regards, Deon ___ 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] chan_oh323: __use_ast_pthread_create_instead __ (was: chan_oh323 loading error)
I'm trying to make the chan_h323 in /usr/src/asterisk/channels/h323
But I'm getting all kinds of errors about PWLIB... I built using the newest
PWLIB and OpenH323 from CVS
Error log from make below
make
g++ -g -c -fno-rtti -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN
-DNDEBUG -DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC -DP_LINUX -D_REENTRANT
-D_GNU_SOURCE -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES
-DPTRACING -DP_USE_PRAGMA -I../../include -I/root/pwlib/include/ptlib/unix
-I/root/pwlib/include -I/root/openh323/include -Wno-missing-prototypes
-Wno-missing-declarations ast_h323.cpp
In file included from /root/pwlib/include/ptlib.h:154,
from ast_h323.cpp:34:
/root/pwlib/include/ptbuildopts.h:157:1: warning: P_LINUX redefined
ast_h323.cpp:1:1: warning: this is the location of the previous definition
In file included from /root/pwlib/include/ptlib.h:178,
from ast_h323.cpp:34:
/root/pwlib/include/ptlib/unix/ptlib/pdirect.h:78: parse error before `
protected'
/root/pwlib/include/ptlib/unix/ptlib/pdirect.h:80: syntax error before `*'
token
In file included from /root/pwlib/include/ptlib.h:190,
from ast_h323.cpp:34:
/root/pwlib/include/ptlib/unix/ptlib/config.h:53: parse error before
`public'
/root/pwlib/include/ptlib/unix/ptlib/config.h:55: destructors must be member
functions
/root/pwlib/include/ptlib/unix/ptlib/config.h:57: parse error before
`protected
'
In file included from /root/pwlib/include/ptlib.h:196,
from ast_h323.cpp:34:
/root/pwlib/include/ptlib/args.h:121: parse error before `{' token
/root/pwlib/include/ptlib/args.h:147: parse error before `const'
/root/pwlib/include/ptlib/args.h:156: parse error before `const'
/root/pwlib/include/ptlib/args.h:165: parse error before `int'
/root/pwlib/include/ptlib/args.h:175: parse error before `int'
/root/pwlib/include/ptlib/args.h:191: virtual outside class declaration
/root/pwlib/include/ptlib/args.h:191: non-member function `void
PrintOn(std::ostream)' cannot have `const' method qualifier
/root/pwlib/include/ptlib/args.h:198: virtual outside class declaration
/root/pwlib/include/ptlib/args.h:206: parse error before `' token
/root/pwlib/include/ptlib/args.h:215: parse error before `' token
/root/pwlib/include/ptlib/args.h:246: virtual outside class declaration
/root/pwlib/include/ptlib/args.h:249: parse error before `' token
/root/pwlib/include/ptlib/args.h:254: virtual outside class declaration
/root/pwlib/include/ptlib/args.h:266: virtual outside class declaration
/root/pwlib/include/ptlib/args.h:266: non-member function `PINDEX
GetOptionCount(char)' cannot have `const' method qualifier
/root/pwlib/include/ptlib/args.h:270: virtual outside class declaration
/root/pwlib/include/ptlib/args.h:270: non-member function `PINDEX
GetOptionCount(const char*)' cannot have `const' method qualifier
/root/pwlib/include/ptlib/args.h:273: parse error before `' token
/root/pwlib/include/ptlib/args.h:274: virtual outside class declaration
/root/pwlib/include/ptlib/args.h:274: non-member function `PINDEX
GetOptionCount(...)' cannot have `const' method qualifier
/root/pwlib/include/ptlib/args.h:283: non-member function `BOOL
HasOption(char)
' cannot have `const' method qualifier
/root/pwlib/include/ptlib/args.h:287: non-member function `BOOL
HasOption(const
char*)' cannot have `const' method qualifier
/root/pwlib/include/ptlib/args.h:290: parse error before `' token
/root/pwlib/include/ptlib/args.h:291: non-member function `BOOL
HasOption(...)'
cannot have `const' method qualifier
/root/pwlib/include/ptlib/args.h:301: syntax error before `(' token
/root/pwlib/include/ptlib/args.h:306: syntax error before `(' token
/root/pwlib/include/ptlib/args.h:311: syntax error before `(' token
/root/pwlib/include/ptlib/args.h:323: non-member function `PINDEX
GetCount()'
cannot have `const' method qualifier
/root/pwlib/include/ptlib/args.h:330: parse error before `=' token
/root/pwlib/include/ptlib/args.h:339: parse error before `)' token
/root/pwlib/include/ptlib/args.h:347: syntax error before `operator'
/root/pwlib/include/ptlib/args.h:363: `PArgList operator(int)' must have
an
argument of class or enumerated type
/root/pwlib/include/ptlib/args.h:363: `PArgList operator(int)' must take
exactly two arguments
/root/pwlib/include/ptlib/args.h:370: `PArgList operator(int)' must have
an
argument of class or enumerated type
/root/pwlib/include/ptlib/args.h:370: `PArgList operator(int)' must take
exactly two arguments
/root/pwlib/include/ptlib/args.h:381: virtual outside class declaration
/root/pwlib/include/ptlib/args.h:381: non-member function `void
IllegalArgumentIndex(int)' cannot have `const' method qualifier
/root/pwlib/include/ptlib/args.h:388: parse error before `' token
/root/pwlib/include/ptlib/args.h:389: virtual outside class declaration
/root/pwlib/include/ptlib/args.h:389: non-member function `void
UnknownOption(...)'
Re: [Asterisk-Users] Advice on BT ISDN Services (UK)
On Wed, 25 Aug 2004 14:03:36 +0100, Jon Fautley [EMAIL PROTECTED] wrote: On 25 Aug 2004, at 13:42, Benjamin Johnson wrote: Thanks for that Jon, can anyone confirm whether Asterisk can pick up which MSN has been dialed and route the call depending on this - or does this functionality only work for DDIs. If I have to use DDIs can anyone recommend and active ISDN card which works with Asterisk and is readily available in the UK. I use a BT Speedway card and chan_capi under * with MSN's works fine no issues. Jason ___ 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] AGI and EXEC function CONFIRMATION
Howdie- Can anyone please confirm that the BACKGROUND application does ***NOT*** return IMMEDIATELY when called from within an AGI EXEC command? It seems that EXEC waits until DTMF or THE END OF THE AUDIO FILE to return to the AGI script. This essentially prevents repeated calls to BACKGROUND (or I assume any other asterisk dialplan application) from within an AGI, and prevents building a queue of audio files--- which is the default functionality of launching ***SEQUENTIAL*** BACKGROUND entries from the DIALPLAN. Is there a work-around from within AGI which will return IMMEDIATELY? The practical application here is to queue-up the following INDIVIDUAL audio segments from within the AGI and go on to other tasks without EXEC (or is it AGI command?) blocking the building of the audio queue: Build Queue with calls to EXEC Background-- segment 1: Option 1 segment 2: Press 11 for Sales segment 3: Option 2 segment 4: Press 22 for Marketing ...etc Return and do other stuff IMMEDIATELY. Thanks Cheers- JJQ _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ 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] Re: Overhead Paging
Quoting Brian Pavane: My plan is to connect a Paging Unit to an FXS port of an IAD, and assign an -Brian Not sure what a Paging Unit is. Some kind of auto-answer phone with audio outputs?? I just used the sound card in the PC plugged into an amplifier. Haven't seen any detrimental effects using the local processing power for this. [paging] ; Overhead paging through the sound card exten = 2900,1,Ringing exten = 2900,2,Dial,console/dsp exten = 2900,3,Hangup -- David Cook ___ 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] Which end hungup?
On Wed, 25 Aug 2004 19:38:34 +0100 (BST), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I suspect it's the POTS end since I haven't been able to reproduce it by dialling IAXComm from a SIP client connected to Asterisk 1, but I can't confirm it. What would cause the X100P to randomly drop a call if this is the case? Some sound during the conversation the card has detected as a busy tone set busydetect=no in zapata.conf or increase the busycount=4 to a higer value, if you need busy detection. Jason ___ 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't make zaptel on red hat 9
Hello, I've followed the instructions here: http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation, but I get the following when trying to make zaptel: I've added the relevant symlinks: [EMAIL PROTECTED] src]# ls -ld /usr/src/linux* lrwxrwxrwx1 root root 14 Aug 26 22:50 /usr/src/linux - linux-2.4.20-8 lrwxrwxrwx1 root root 14 Aug 27 2004 /usr/src/linux-2.4 - linux-2.4.20-8 drwxr-xr-x 16 root root 4096 Aug 26 22:59 /usr/src/linux-2.4.20-8 How do I get zaptel installed? [EMAIL PROTECTED] zaptel]# 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-point er -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include /usr/src/linux-2.4/include/linux/modversions.h -DSTANDALONE_ZAPATA -c zaptel.c cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file 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-point er -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include /usr/src/linux-2.4/include/linux/modversions.h -DSTANDALONE_ZAPATA -c tor2.c In file included from tor2.c:30: /usr/src/linux-2.4/include/linux/kernel.h:60: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:60: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:60: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:61: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:61: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:62: `panic_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:62: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:68: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:68: `simple_strtoul_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:68: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:69: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:69: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:69: `simple_strtol_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:69: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:70: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:70: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:70: `simple_strtoull_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:70: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:72: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:72: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:73: `sprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:73: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:74: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:74: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:74: `vsprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:74: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:75: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:75: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:76: `snprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:76: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:77: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:77: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:77: `vsnprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:77: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:79: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:79: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:80: `sscanf_R_ver_str' declared as function returning a function
Re: [Asterisk-Users] Asterisk PBX and backup Circuits
I am interested to know how one would calculate the amount of PSTN connection needed for backup on an Asterisk PBX that is being setup to receive its DIDs via a VoIP provide. To sum up what I am implementing: I am porting my DIDs to a VoIP provide so I will need a back up plan in place if the Data network fails. In addition, 911 will always be going out the PSTN so I know I need at least one POTs circuit. Calls inbound and outbound will always routed through the data network. In telco terms, you probably need to do a small Traffic Study; analyze the existing traffic for maximum number of simultanous calls, etc. If this is an existing business with an existing pbx, there are likely some usage statistics available within the pbx. If that's not available, some telephone companies will do the traffic study for you (don't need to tell them why your doing it, but rather to determine the number of telco lines needed for the business.) If that's not possible, ask the telco to provide you with a list of all calls with detail and run through the list to calculate the maximum number of simultanous calls. If this is a new installation with absolutely no history, your only option is to guess at the maximum. ___ 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] Overhead Paging
I am currently implementing a VoIP PBX, and need to deal with the paging situation. I would prefer to do paging via overhead speakers. My plan is to connect a Paging Unit to an FXS port of an IAD, and assign an extension to that port. I would then simply be able to call that extension, and have my call patched through to the overhead speakers. Has anyone implemented this type of setup, if so, what type of paging unit did you deploy, did you require an external amplifier or power supply, and how many speakers were you able to connect to the unit? As it stands, I will need between 4 and 8 speakers, and some of the speakers will be 400 feet from the main telco closet. Any thoughts, comments, and suggestions that you can shed on this topic would be much appreciated. If you have other methods of implementing overhead paging, I would also be interested. If you search the archives I think you'll find this discussed several times. One (of many) ways to accomplish it is simply based on using a Cisco 7940/7960 phone configured with paging, and pipe the audio to an amplifier input. If you're planning on deploying the Cisco phones already, then using that approach basically has built-in sparing covered. ___ 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] chan_oh323 build (resubmit w/ new title)
I'm trying to make the chan_h323 in /usr/src/asterisk/channels/h323
But I'm getting all kinds of errors about PWLIB... I built using the newest
PWLIB and OpenH323 from CVS
Error log from make below
make
g++ -g -c -fno-rtti -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN
-DNDEBUG -DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC -DP_LINUX -D_REENTRANT
-D_GNU_SOURCE -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES
-DPTRACING -DP_USE_PRAGMA -I../../include -I/root/pwlib/include/ptlib/unix
-I/root/pwlib/include -I/root/openh323/include -Wno-missing-prototypes
-Wno-missing-declarations ast_h323.cpp
In file included from /root/pwlib/include/ptlib.h:154,
from ast_h323.cpp:34:
/root/pwlib/include/ptbuildopts.h:157:1: warning: P_LINUX redefined
ast_h323.cpp:1:1: warning: this is the location of the previous definition
In file included from /root/pwlib/include/ptlib.h:178,
from ast_h323.cpp:34:
/root/pwlib/include/ptlib/unix/ptlib/pdirect.h:78: parse error before `
protected'
/root/pwlib/include/ptlib/unix/ptlib/pdirect.h:80: syntax error before `*'
token
In file included from /root/pwlib/include/ptlib.h:190,
from ast_h323.cpp:34:
/root/pwlib/include/ptlib/unix/ptlib/config.h:53: parse error before
`public'
/root/pwlib/include/ptlib/unix/ptlib/config.h:55: destructors must be member
functions
/root/pwlib/include/ptlib/unix/ptlib/config.h:57: parse error before
`protected
'
In file included from /root/pwlib/include/ptlib.h:196,
from ast_h323.cpp:34:
/root/pwlib/include/ptlib/args.h:121: parse error before `{' token
/root/pwlib/include/ptlib/args.h:147: parse error before `const'
/root/pwlib/include/ptlib/args.h:156: parse error before `const'
/root/pwlib/include/ptlib/args.h:165: parse error before `int'
/root/pwlib/include/ptlib/args.h:175: parse error before `int'
/root/pwlib/include/ptlib/args.h:191: virtual outside class declaration
/root/pwlib/include/ptlib/args.h:191: non-member function `void
PrintOn(std::ostream)' cannot have `const' method qualifier
/root/pwlib/include/ptlib/args.h:198: virtual outside class declaration
/root/pwlib/include/ptlib/args.h:206: parse error before `' token
/root/pwlib/include/ptlib/args.h:215: parse error before `' token
/root/pwlib/include/ptlib/args.h:246: virtual outside class declaration
/root/pwlib/include/ptlib/args.h:249: parse error before `' token
/root/pwlib/include/ptlib/args.h:254: virtual outside class declaration
/root/pwlib/include/ptlib/args.h:266: virtual outside class declaration
/root/pwlib/include/ptlib/args.h:266: non-member function `PINDEX
GetOptionCount(char)' cannot have `const' method qualifier
/root/pwlib/include/ptlib/args.h:270: virtual outside class declaration
/root/pwlib/include/ptlib/args.h:270: non-member function `PINDEX
GetOptionCount(const char*)' cannot have `const' method qualifier
/root/pwlib/include/ptlib/args.h:273: parse error before `' token
/root/pwlib/include/ptlib/args.h:274: virtual outside class declaration
/root/pwlib/include/ptlib/args.h:274: non-member function `PINDEX
GetOptionCount(...)' cannot have `const' method qualifier
/root/pwlib/include/ptlib/args.h:283: non-member function `BOOL
HasOption(char)
' cannot have `const' method qualifier
/root/pwlib/include/ptlib/args.h:287: non-member function `BOOL
HasOption(const
char*)' cannot have `const' method qualifier
/root/pwlib/include/ptlib/args.h:290: parse error before `' token
/root/pwlib/include/ptlib/args.h:291: non-member function `BOOL
HasOption(...)'
cannot have `const' method qualifier
/root/pwlib/include/ptlib/args.h:301: syntax error before `(' token
/root/pwlib/include/ptlib/args.h:306: syntax error before `(' token
/root/pwlib/include/ptlib/args.h:311: syntax error before `(' token
/root/pwlib/include/ptlib/args.h:323: non-member function `PINDEX
GetCount()'
cannot have `const' method qualifier
/root/pwlib/include/ptlib/args.h:330: parse error before `=' token
/root/pwlib/include/ptlib/args.h:339: parse error before `)' token
/root/pwlib/include/ptlib/args.h:347: syntax error before `operator'
/root/pwlib/include/ptlib/args.h:363: `PArgList operator(int)' must have
an
argument of class or enumerated type
/root/pwlib/include/ptlib/args.h:363: `PArgList operator(int)' must take
exactly two arguments
/root/pwlib/include/ptlib/args.h:370: `PArgList operator(int)' must have
an
argument of class or enumerated type
/root/pwlib/include/ptlib/args.h:370: `PArgList operator(int)' must take
exactly two arguments
/root/pwlib/include/ptlib/args.h:381: virtual outside class declaration
/root/pwlib/include/ptlib/args.h:381: non-member function `void
IllegalArgumentIndex(int)' cannot have `const' method qualifier
/root/pwlib/include/ptlib/args.h:388: parse error before `' token
/root/pwlib/include/ptlib/args.h:389: virtual outside class declaration
/root/pwlib/include/ptlib/args.h:389: non-member function `void
UnknownOption(...)'
[Asterisk-Users] ISDN Card Recommendation
I'm running Asterisk 1.0 RC2 on a RedHat 9.0 box. I have a ISDN BRI line that I would like hook up to my Asterisk server and would like to ask the group what you guys would recommend as far as isdn cards that install easily into the Linux and asterisk environment. Rgs, Mark ___ 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] Overhead Paging
Valcom makes commerical paging subsystems for PBX's and I see that they have made a VoIP compatible pageing module. Lyle - Original Message - From: Brian Pavane [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, August 26, 2004 7:12 AM Subject: [Asterisk-Users] Overhead Paging All, I am currently implementing a VoIP PBX, and need to deal with the paging situation. I would prefer to do paging via overhead speakers. My plan is to connect a Paging Unit to an FXS port of an IAD, and assign an extension to that port. I would then simply be able to call that extension, and have my call patched through to the overhead speakers. Has anyone implemented this type of setup, if so, what type of paging unit did you deploy, did you require an external amplifier or power supply, and how many speakers were you able to connect to the unit? As it stands, I will need between 4 and 8 speakers, and some of the speakers will be 400 feet from the main telco closet. Any thoughts, comments, and suggestions that you can shed on this topic would be much appreciated. If you have other methods of implementing overhead paging, I would also be interested. -Brian ___ 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] Codec
On Thursday 26 August 2004 04:13, altus wrote: I want to know what the best codec is to use for asteris for VOIP We have two towns connected with a 64k line that's going to do VOIP with astersik.At the moment with the default installation the quality is bad and the bandwith is high. Is this even a codec problem 64kbps is not a lot of bandwidth; You would probably be best with g729 (it costs $10 per simultaneous transcode) -- i.e. if you only ever have one conversation at a time then it's $10 for your license. You can also try iLBC, speex or GSM but you're rapidly running into the limit of your link. If you don't care about audio quality you can play with LPC10. It's extremely low bandwidth but if you try it you'll know why. :-) There is no best codec -- it's all a balance of budget, quality and bandwidth. -A. ___ 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] ISDN Card Recommendation
Hi Mark, I've succesfully installed an Eicon Diva 2.0 PCI in 10 minutes. It works fine but i have some problems with CALLER NUMBER: it's always 0. My box is RedHat 9.0 and asterisk cvs 25-08-2004. Bye -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paterson, Mark Sent: giovedì 26 agosto 2004 15.26 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ISDN Card Recommendation I'm running Asterisk 1.0 RC2 on a RedHat 9.0 box. I have a ISDN BRI line that I would like hook up to my Asterisk server and would like to ask the group what you guys would recommend as far as isdn cards that install easily into the Linux and asterisk environment. Rgs, Mark ___ 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] spandsp and certain (e.g. Canon) fax machines
On Wednesday 25 August 2004 23:50, Steve Underwood wrote: Several people have reported problems sending faxes from spandsp-0.0.1k to Canon FAX machines. A spandsp user had the same problem with another make of FAX machine, and traced the problem to a bug in the file t30.c of spandsp. Line 542 says s-t4.rx_file[0] where it should say s-t4.tx_file[0]. This fixes his problem, and I suspect it will also fix the Canon fax machine problem. Can someone having problems with Canon machines try this change, and tell me the result? I will give this a shot shortly. I still get spandsp segfaulting the odd time so I need to set up a secondary asterisk box to prevent such problems. I've already posted to the list about that particular problem, it's not something as simple as the wrong copy of libtiff or anything. :-) Just FYI; we have a Canon IR3300 fax/copier/scanner big badass unit (over 3mil copies and going...) -A. ___ 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] chan_sccp2 7960 -- documentation and example request.
I'm guessing that chan_sccp2 is the same one I am using which was downloaded from http://chan-sccp.sourceforge.net ? If it is, we are in luck. I have 2 Cisco 7960G's all running just fine with this new module. 1 phone has 1 line on it and the other has 2 lines on it. I am able to dial all 3 lines from both phones and can call POTS numbers as well as all our SIP phones. I've got a custom services page running (hosted by someone else) and are in the process of getting a custom directory working. Speeddials also work great displaying my custom name on the LCD screen. Have not yet figured out how to put call on hold but can transfer calls to another extension and can park calls. I also applied a patch to the module allowing a multi-line phone to answer an incomming call on any line. The non-patched version of CVS does not do this. Certain softkey menus (the 4 buttons along the bottom of the LCD) do not seem to be visible in the correct mode. For example, I can see a 'Hold' option visible right now even though there is no active call. And the 'Hold' button dissapears when a call is active; so I can't press 'Hold' or 'Transfer'. But I can use *'s internal #EXT to transfer. Haven't tested intercom abilities yet. Make sure that in your modules.conf you have a noload = chan_skinny.so otherwise the 7960's will continue to use that module instead of the new chan_sccp. sccp.conf - [general] keepalive = 60 ; How often the SCCP device does a keepalive ping context = default ; default context that will be used if nothing else is specified for dateFormat = D-M-Y ; M-D-Y in any order (5 chars max) bindaddr = 1.2.3.4 ; replace 1.2.3.4 with the ip address of the asterisk box. port = 2000 ; listen on port 2000 (Skinny, default) [SEP000F3442E4A7] description = Matthew's 7960; A description, may be up to 16 charecters long. Used by * in 'sccp show' type = 7960 ; The model type needs to be defined so we know how to set it up. context = matthew ; default context for outgoing calls. tzoffset = -6 ; Timezone offset from GMT autologin = 1001 speeddial = 4,John Doe speeddial = 7,Jack Trades speeddial = 8,Neverwinter Nights [SEP000F3442E199] description = Jack's 7960 type = 7960 context = matthew tzoffset = -6 autologin = 1002,1003 speeddial = 4,John Doe speeddial = 7,Jack Trades speeddial = 8,Neverwinter Nights [1001] id = 1001 ; Id is a number that is dialed to login to the line with. pin = 1234 ; The pin number needed to log into the device. If pin is missing, anyone can log into it label = 1001 ; The text to display on the display (on 7960) description = 1001 ; The text to display on the screen (on the 7910) context = matthew ; Context outgoign calls are in. callwaiting = 1 ; If set to 1, call waiting will work. ;mailbox = 4; Check if this mailbox has any mail, and if so, show the Message Waiting Indicator. callerid= Theo 1001 ; CallerId to use on outgoing calls from this line. [1002] id = 1002 pin = 1234 label = 1002 description = 1002 context = matthew callwaiting = 1 ;mailbox = 5 callerid= Richard 1002 [1003] id = 1003 pin = 1234 label = 1003 description = 1003 context = matthew callwaiting = 1 ;mailbox = 6 callerid= Neill 1003 extensions.conf [matthew] exten = 1001,1,Dial(SCCP/1001,15,tr) exten = 1002,1,Dial(SCCP/1002,15,tr) exten = 1003,1,Dial(SCCP/1003,15,tr) Hope this helps some. Matthew - Original Message - From: Paterson, Mark [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 10:01 PM Subject: RE: [Asterisk-Users] chan_sccp2 7960 -- documentation andexample request. I would be very interested in this as well. Rgs, Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Lawrence Sent: Wednesday, August 25, 2004 9:57 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_sccp2 7960 -- documentation and example request. Can someone provide example configuration files and/or instructions for configuring Asterisk to work with chan_sccp2 and a 7960. I have the searched the chan_sccp2 site, the voip_info wiki site, asterisk doc project, and mailing list archives for the past year for help, but failed to find anything helpful. Some of the questions I have are: 1. What settings need to be made on the phone? 2. What configuration files are required? 3. What is the syntax/format of the configuration files? 4. What is the syntax for Dial() when using sccp2? I have no experience with skinny/sccp as I have been using SIP and IAX2. I would like to try to take advantage of the extended featureset of the 7960s in SCCP mode. Any help would be GREATLY appreciated. Thanks! Robert Lawrence
Re: [Asterisk-Users] spandsp and certain (e.g. Canon) fax machines
On Thu, 2004-08-26 at 09:38, Andrew Kohlsmith wrote: On Wednesday 25 August 2004 23:50, Steve Underwood wrote: Several people have reported problems sending faxes from spandsp-0.0.1k to Canon FAX machines. A spandsp user had the same problem with another make of FAX machine, and traced the problem to a bug in the file t30.c of spandsp. Line 542 says s-t4.rx_file[0] where it should say s-t4.tx_file[0]. This fixes his problem, and I suspect it will also fix the Canon fax machine problem. Can someone having problems with Canon machines try this change, and tell me the result? I will give this a shot shortly. I still get spandsp segfaulting the odd time so I need to set up a secondary asterisk box to prevent such problems. I've already posted to the list about that particular problem, it's not something as simple as the wrong copy of libtiff or anything. :-) Just FYI; we have a Canon IR3300 fax/copier/scanner big badass unit (over 3mil copies and going...) -A. I too have an ir3300 and was having issues with faxes. I found this googleing: Hooper: IR330 w/ print and fax works just fine. Except that it does not receive from one customer. May I add that it's their most important customer! The faxes are coming from a computer. I have looked at my past note on a similar problem and did this: SSW3 1000-0010 from - SSW17 -0010 from - MEM NL to on from off ATT to 6 NUM 02 to 15 03 to 20 04 to 15 010 to 6800 Not sure of the rom versions. Still isn't receiving. Any suggestions? RussW Try setting SW05 bit 3 to 1 from 0 I have seen this same fault with the HP31xx Hooper: Problem fixed! Changed SSW5 bit three to 1 from zero. I only changed the ssw5 bit the other changes seemed to make it worse. Hope this helps! t o n y ___ 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] AGI and EXEC function CONFIRMATION
At least with my php scripts, it retuns immediately, which is why i get it to check for what the function returns eg. fputs($configSettings['handles']['stdOut'],STREAM FILE $filename \\\n); gets it to play a background message This line keeps my script from continueing (and thus, ending and preventing asterisk skipping) $temp = fgets($configSettings['handles']['stdIn']); hope that helps Try waiting until jeff quade wrote: Howdie- Can anyone please confirm that the BACKGROUND application does ***NOT*** return IMMEDIATELY when called from within an AGI EXEC command? It seems that EXEC waits until DTMF or THE END OF THE AUDIO FILE to return to the AGI script. This essentially prevents repeated calls to BACKGROUND (or I assume any other asterisk dialplan application) from within an AGI, and prevents building a queue of audio files--- which is the default functionality of launching ***SEQUENTIAL*** BACKGROUND entries from the DIALPLAN. Is there a work-around from within AGI which will return IMMEDIATELY? The practical application here is to queue-up the following INDIVIDUAL audio segments from within the AGI and go on to other tasks without EXEC (or is it AGI command?) blocking the building of the audio queue: Build Queue with calls to EXEC Background-- segment 1: Option 1 segment 2: Press 11 for Sales segment 3: Option 2 segment 4: Press 22 for Marketing ...etc Return and do other stuff IMMEDIATELY. Thanks Cheers- JJQ _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ 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: Problems compiling chan_h323 (was: [Asterisk-Users] chan_oh323: __use_ast_pthread_create_instead ...)
Huddleston, Robert schrieb: I'm trying to make the chan_h323 in /usr/src/asterisk/channels/h323 But I'm getting all kinds of errors about PWLIB... I built using the newest PWLIB and OpenH323 from CVS Error log from make below ... Please start a new thread next time when asking about a new topic! Hi, please have a look into the README file inside channels/h323! Maybe that has changed during the past days, but some days ago h323 had to be compiled with: - openh323-1.12.2 - pwlib-1.5.2 whereas chan_oh323 currently has to be compiled with - openh323-1.13.5 - pwlib-1.6.6 and the patch included in asterisk-oh has to be applied to openh323-1.13.5/ Don't use CVS-versions in order to compile those asterisk- channels! Roger. ___ 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] ISDN Card Recommendation
Good luck finding a Linux/Asterisk-compatible ISDN PCI card that's usable in the US. I've had my eye out for a while, and haven't had any luck. If you do find something, and find someone who will sell it to you, please share it with the list... Rob On Thu, 26 Aug 2004 08:26:24 -0500, Paterson, Mark [EMAIL PROTECTED] wrote: I'm running Asterisk 1.0 RC2 on a RedHat 9.0 box. I have a ISDN BRI line that I would like hook up to my Asterisk server and would like to ask the group what you guys would recommend as far as isdn cards that install easily into the Linux and asterisk environment. Rgs, Mark ___ 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] Astricon hotel recommendations.....?
Hi all, Looks like I'm going to make the trip over to Astricon next month, but finances being what they are and since I'll be paying for my own flight from the UK, I'm trying to cut down on costs. The problem I've got is that the hotel is about 4 miles from the nearest public transportation (Brookhaven station) and the shuttle bus is only for hotel residents. Hiring a car seems a little like overkill (and expensive) when the public transport seems to be very good. So... does anybody have any recommendations for a cheap hotel/motel I could stay in? Also, if I have to get a taxi/cab, how much is it likely to cost (in $ per mile)? Is anybody else staying elsewhere and would be willing to share transport costs? Cheers, Nick Barnes ___ 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] Which end hungup?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thu, 26 Aug 2004, Jason Williams wrote: Some sound during the conversation the card has detected as a busy tone set busydetect=no in zapata.conf or increase the busycount=4 to a higer value, if you need busy detection. I'm not using busy detection. - -- - Steve Jabber: [EMAIL PROTECTED] Web: http://www.nexusuk.org/ Servatis a periculum, servatis a maleficum - Whisper, Evanescence -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Public key available at http://www.nexusuk.org/pubkey.txt iD8DBQFBLfFV5zUOsIV3bqERAjzEAKCPD/BJhb5crMTLoUCfNJNML/tvUQCfZOIQ QrOodbiNtJ1Hu0xUIxAiKV0= =kePB -END PGP SIGNATURE- ___ 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] Which end hungup?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thu, 26 Aug 2004 [EMAIL PROTECTED] wrote: Some sound during the conversation the card has detected as a busy tone set busydetect=no in zapata.conf or increase the busycount=4 to a higer value, if you need busy detection. I'm not using busy detection. My mistake - yes I am (not sure why since it doesn't work here in the UK). I've turned it off now so I'll see if that helps. Thanks. - -- - Steve Jabber: [EMAIL PROTECTED] Web: http://www.nexusuk.org/ Servatis a periculum, servatis a maleficum - Whisper, Evanescence -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Public key available at http://www.nexusuk.org/pubkey.txt iD8DBQFBLfIy5zUOsIV3bqERAu/fAKCSzgv1FEvIL8B57pXWCDf73iNuxACdGDzJ SznQ1vvZA379rgGNJvAq2Mc= =YkYl -END PGP SIGNATURE- ___ 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] Astricon hotel recommendations.....?
I wrote: Looks like I'm going to make the trip over to Astricon next month, but finances being what they are and since I'll be paying for my own flight from the UK, I'm trying to cut down on costs. I should add that in order to cut the cost of the air fare by 80% (yes, 80%), I have to stay over longer and depart on the Sunday, so I'll need 5 nights of accommodation, hence my need to keep it cheap - I can't afford $111 x 5. Cheers, Nick Barnes. ___ 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] chan_sccp2 7960 -- documentation and example request.
Matthew Boehm ([EMAIL PROTECTED]) wrote: Matthew: See my comments inline... I'm guessing that chan_sccp2 is the same one I am using which was downloaded from http://chan-sccp.sourceforge.net ? yep it is. If it is, we are in luck. I have 2 Cisco 7960G's all running just fine with this new module. Good to hear :-) 1 phone has 1 line on it and the other has 2 lines on it. I am able to dial all 3 lines from both phones and can call POTS numbers as well as all our SIP phones. I've got a custom services page running (hosted by someone else) and are in the process of getting a custom directory working. Speeddials also work great displaying my custom name on the LCD screen. We'll soon will modify to easy speeddial handling (e.g. handling in memory). Have not yet figured out how to put call on hold but can transfer calls to another extension and can park calls. On Hold should work by using the button, but haven't extensively tested it yet on the 7960G I also applied a patch to the module allowing a multi-line phone to answer an incomming call on any line. The non-patched version of CVS does not do this. I'm currently incorporating this patch into CVS as it seems useful to me and others. Certain softkey menus (the 4 buttons along the bottom of the LCD) do not seem to be visible in the correct mode. For example, I can see a 'Hold' option visible right now even though there is no active call. And the 'Hold' button dissapears when a call is active; so I can't press 'Hold' or 'Transfer'. But I can use *'s internal #EXT to transfer. okay, i'll check this, this seems to be an easy to solve issue Haven't tested intercom abilities yet. Don't try this at home :-) AFAIK this is not supported neither correctly programmed, but after the softkeys / speeddials are finished, i'll work on the intercom stuff. --jan -- Jan Czmok, Network Engineering Support, Global Access Telecomm, Inc. Ph.: +49 69 299896-35 - fax: +49 69 299896-40 - sip:[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] Astricon hotel recommendations.....?
I lived there for many years. There should be buses to Brookhaven from most anywhere in the area. Go to 'itsmarta.com' to get schedules or call your hotel and they will be able to help you. Nick Barnes wrote on 8/26/04, 9:15 AM: Hi all, Looks like I'm going to make the trip over to Astricon next month, but finances being what they are and since I'll be paying for my own flight from the UK, I'm trying to cut down on costs. The problem I've got is that the hotel is about 4 miles from the nearest public transportation (Brookhaven station) and the shuttle bus is only for hotel residents. Hiring a car seems a little like overkill (and expensive) when the public transport seems to be very good. So... does anybody have any recommendations for a cheap hotel/motel I could stay in? Also, if I have to get a taxi/cab, how much is it likely to cost (in $ per mile)? Is anybody else staying elsewhere and would be willing to share transport costs? Cheers, Nick Barnes ___ 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] Wil spandsp work with i4l driver?
Hi, will spandsp work with i4l, or should I use capi? tia mazek -- http://www.marcinmazurek.com/ ::: nic-hdl: MM3380-RIPE GnuPG 6687 E661 98B0 AEE6 DA8B 7F48 AEE4 776F 5688 DC89 ___ 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] ilbc asterisk and handytone/Xpro
Have a small asterisk implementation using xten xpro and handytone 286 clients Currently using ulaw, would like to use ilbc for bandwidth reason Can anyone help me with settings i will need in asterisk and clients for this to work Thanks ___ 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] chan_oh323 and cdr
Roger Schreiter wrote: Hi, there are some posts about that topic, but unfortunatelly I do not yet know what to do. I find every call in Master.csv, but those coming in via chan_oh323. In oh323.conf I have accountcode=oh323 but there is no other file in the directory cdr-csv than Master.csv. Can anyone give me any hint, what to do, in order to have calls from chan_oh323 logged in any file? In oh323.conf, set: amaFlags=billing Michael. ___ 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 and Eicon Diva 2.0
Hi all. This is my first post, but i don't know if i'm off topic. I've a linux box with RedHat 9.0 (linux kernel 2.4.8-20) and asterisk cvs installed on it, Eicon Diva 2.0 and a BRI ISDN line in italy. Everything works fine but i can't handle the CALLER NUMBER: it's alway set to 0. Can someone in Italy help me ? Thanks in adcance. ___ 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] RC2 and Netmeeting 3.01 ?
Zineddin Karzazi wrote: --- Robert Rozman [EMAIL PROTECTED] schrieb: Hi, I'd kindly ask for any guidance how to setup Netmeeting to work with Asterisk. I've setup Asterisk as Gateway, selected GSM codec, and I'm able to call local extensions (no calls into PBX functions) but get no sound. Any hint, advice ? Anyone using Netmeeting (maybe also windows messenger) with Asterisk sucessfully ? Thanks in advance, regards, Robert. I have same Problems with Netmeeting, just wanted to test H.323 with Astersik , it rings, but as soon as i answer it dissconnects. im getting the Following Error: oh323_exception: OH323/R27469: Invalid format of RTP addresses. Aug 13 10:19:05 ERROR[524304]: chan_oh323.c:1933 oh323_write: OH323/R27469: Failed to create smoother. There is no common codec between Asterisk and Netmeeting. Today i tried Openphone (H.323 Client) and it works. Zineddin. Michael. ___ 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+IVR functions trouble
I' got a problem, using asterisk-rc2 :IVR functions (Background...Playback...etc) doesn't works : Executing Background(OH323/RX, vm-extension) in new stack channel.c:1650 ast_set_write_fornat: Unable to find path from GSM to G729A---Asterisk box supplied only with network adapter.---Asterisk box registered in Mera (soft-switch with H323 protocol) and doing SIP-endpoints (such as ATA). and G729A is preferred codec to my needs.Is this trouble associated with G729A codec? ___ 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] Overhead Paging
I am currently implementing a VoIP PBX, and need to deal with the paging situation. I would prefer to do paging via overhead speakers. My plan is to connect a "Paging Unit" to an FXS port of an IAD, and assign an extension to that port. I would then simply be able to call that extension, and have my call patched through to the overhead speakers. Has anyone implemented this type of setup, if so, what type of paging unit did you deploy, did you require an external amplifier or power supply, and how many speakers were you able to connect to the unit? As it stands, I will need between 4 and 8 speakers, and some of the speakers will be 400 feet from the main telco closet. Any thoughts, comments, and suggestions that you can shed on this topic would be much appreciated. If you have other methods of implementing overhead paging, I would also be interested. If you search the archives I think you'll find this discussed several times. One (of many) ways to accomplish it is simply based on using a Cisco 7940/7960 phone configured with paging, and pipe the audio to an amplifier input. If you're planning on deploying the Cisco phones already, then using that approach basically has built-in sparing covered. You can do these with the Grandstream Budgetone. They only cost around $75.00 ___ 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] Echo on polycom sip phone
Title: Echo on polycom sip phone I have 2 of the X100P cards in my Mandrake 9.1 box, and 3 Polycom 500 phones. I have a terrible echo problem. It will be fine and then while you are talking it gets really loud and distorted and then will die down again. The machine is a Duron 750 with 128 MB ram, is that enough? I am hoping that its my machine and upgrading will solve it. But if not, then I have a real problem. HELP! Thanks Sean Garland Siskiyou Technology Consultants s ___ 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] Astricon hotel recommendations.....?
Rick L. Wilson, Sr. wrote: I lived there for many years. There should be buses to Brookhaven from most anywhere in the area. Sorry, I probably didn't do a good job of explaining... The problem is that the Mariott is 4 miles from Brookhaven, there doesn't seem to be any public transport between the hotel and Brookhaven and the hotel shuttle is for hotel residents only. So... How do I get from Brookhaven to the Marriott? Nick Barnes ___ 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+IVR functions trouble
alex3377 wrote: I' got a problem, using asterisk-rc2 :IVR functions (Background...Playback...etc) doesn't works : Executing Background(OH323/RX, vm-extension) in new stack channel.c:1650 ast_set_write_fornat: Unable to find path from GSM to G729A---Asterisk box supplied only with network adapter.---Asterisk box registered in Mera (soft-switch with H323 protocol) and doing SIP-endpoints (such as ATA). and G729A is preferred codec to my needs.Is this trouble associated with G729A codec? Do you have G.729 codec for Asterisk installed? Michael. ___ 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] chan_oh323 and cdr
Michael Manousos schrieb: ... In oh323.conf, set: amaFlags=billing Yes, that solved my problem. Thanks! ___ 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+IVR functions trouble
Hello alex3377, Thursday, August 26, 2004, 6:51:52 PM, you wrote: a channel.c:1650 ast_set_write_fornat: Unable to find path from GSM to G729A You need to purchase G.729 codec license from Digium in order to use G.729 in transcoding mode. http://www.digium.com/index.php?menu=asterisk_g729 P.S.: There is a Russian Asterisk-community site. Visit www.asterisk.org.ru :-) -- 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] need help with zaptel.
Edward, I Forgot to tell you take a look at README.udev in the zaptel directory. Fedora core 2 use udev and you have to make some rules in the /etc/udev Else he will not find the devices /dev/zap/* Han From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Johannes van Hulst Sent: Thursday, August 26, 2004 9:53 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] need help with zaptel. Edward, I had the same problem I am running fedora core 2 with a 2.6.8-1.521 kernel at it is working now perfect. Craig wrote the following : I have had success with this using both the X100p (wcfxs and wcfxo) and TE410p (wct4xxp) under Redhat FC2 2.6.5. The instructions are on the wiki, do the following: ln -s /lib/modules/2.6.5-1.358/build linux-2.6 cd zaptel make clean make linux26 make install I made my softlink to /usr/src/linux-2.6.8-1.521 after that the compiling of Zaptel is working perfectly but you get the inserting module error. After that I changed the softlink to /lib/modules/2.8.9-1.521/build and now everything is working perfectly. But don´t forget to recompile zaptel after changing the softlink Han From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Edward Huitt Sent: Saturday, August 21, 2004 9:22 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] need help with zaptel. Ok I am past the compile. Now when I try to modprobe I get FATAL: Error inserting zaptel (/lib/modules/2.6.5-1.358/misc/zaptel.ko): Invalid module format -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Blakely Sent: Saturday, August 21, 2004 7:00 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] need help with zaptel. Look in /usr/src. You should see a directory something similar to linux-2.6.1-1[as an example]. If you DON'T have a directory (or link to a directory) named linux-2.6, you should create one using the 'ln' command. In the case mentioned above, the command would be: ln -s /usr/src/linux-2.6.1-1 /usr/src/linux-2.6 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Edward Huitt Sent: Saturday, August 21, 2004 5:42 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] need help with zaptel. I cant get zaptel to make. I get this error: make: *** [linux26] Error 1. The previous line is: Link /usr/src/linux-2.6 to your kernel sources first! I am running Fedora Core2 Asterisk compiles fine. I am using my SIP phones. I would like to get my TDM400p working. ___ 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] Echo on polycom sip phone
Please post zapata.conf John Sean Garland wrote: I have 2 of the X100P cards in my Mandrake 9.1 box, and 3 Polycom 500 phones. I have a terrible echo problem. It will be fine and then while you are talking it gets really loud and distorted and then will die down again. The machine is a Duron 750 with 128 MB ram, is that enough? I am hoping that its my machine and upgrading will solve it. But if not, then I have a real problem. HELP! Thanks *Sean Garland* *Siskiyou Technology Consultants* s ___ 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] ISDN Card Recommendation
What's the deal w/ this setup in the US? I'm a little new to the Asterisk thing and my company has an unused ISDN BRI provisioned for both data and voice (NT1) Is this not possible? Rgs, Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Fugina Sent: Thursday, August 26, 2004 9:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ISDN Card Recommendation Good luck finding a Linux/Asterisk-compatible ISDN PCI card that's usable in the US. I've had my eye out for a while, and haven't had any luck. If you do find something, and find someone who will sell it to you, please share it with the list... Rob On Thu, 26 Aug 2004 08:26:24 -0500, Paterson, Mark [EMAIL PROTECTED] wrote: I'm running Asterisk 1.0 RC2 on a RedHat 9.0 box. I have a ISDN BRI line that I would like hook up to my Asterisk server and would like to ask the group what you guys would recommend as far as isdn cards that install easily into the Linux and asterisk environment. Rgs, Mark ___ 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] Error Compiling MySQL Friends
Thanks Flynn but still no go. I looked at line 141 and it seems to be fine. elifeq ($(USE_SIP_MYSQL_FRIENDS),1) I also tried removing the comma, and putting in a tab but I get the same error. I havent made any changes to the file, it was downloaded automatically via cvs. Any other thoughts? - Original Message - From: el Flynn [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, August 26, 2004 1:45 AM Subject: Re: [Asterisk-Users] Error Compiling MySQL Friends imail wrote: All, I edited the Makefile under asterisk/channels and set: USE_MYSQL_FRIENDS=1 USE_SIP_MYSQL_FRIENDS=1 When I do a make clean ; make install I get the following for x in res channels pbx apps codecs formats agi cdr astman stdtime; do make -C $x clean || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk/res' rm -f *.so *.o .depend make[1]: Leaving directory `/usr/src/asterisk/res' make[1]: Entering directory `/usr/src/asterisk/channels' Makefile:141: *** missing separator. Stop. make[1]: Leaving directory `/usr/src/asterisk/channels' make: *** [clean] Error 1 Could it be that your problem is coming from the error: Makefile:141: *** missing separator. Stop. From the rest of the output it doesn't seem to imply there's something missing where MySQL is concerned. Googling on makefile and missing separator gave me this link that may be of help: http://www.cygwin.com/ml/cygwin/2003-07/msg00341.html Although I may be way off, you could try that out first and see if it doesn't solve the problem. Cheers, Flynn ___ 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] Anyone using Asterisk on Slackware 9?
Hi, I am trying to do a very minimal install of Slackware to run Asterisk on. Can anyone give me a list of what packages I need to install as I dont want X an all the associated bloat? Thanks in advance Chris -- ___ 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] Echo on polycom sip phone
Title: Echo on polycom sip phone Hi, We have Polycom IP600 and I fight with the same problem. I call Digium and the guys tell me to play with echocancel and echotraining in Zapata.conf. I use the same X100P for the start and I cannot fix the problem with the echo. Now I get a 4 FXO card and with echotraining=yes and rxwink=800 I have a crystal clear sound. Best regards, Chris HARIGA From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Garland Sent: Thursday, August 26, 2004 10:59 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Echo on polycom sip phone I have 2 of the X100P cards in my Mandrake 9.1 box, and 3 Polycom 500 phones. I have a terrible echo problem. It will be fine and then while you are talking it gets really loud and distorted and then will die down again. The machine is a Duron 750 with 128 MB ram, is that enough? I am hoping that its my machine and upgrading will solve it. But if not, then I have a real problem. HELP! Thanks Sean Garland Siskiyou Technology Consultants s ___ 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] Sound card
Is a sound card needed in order to playback some of the asterisk sounds in /var/lib/asterisk/sounds when dialing out with an X100P? Thanks. ___ 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] Sip Channel CLI
Hello asterisk-users, I have had asterisk registering as a sip extension to an external provider, calls are coming in in pretty fine. Also dialing out works like a charm, the only problem is that calling out asterisk is displayed on the called phone instead of the sip address of the asterisk box. I googled around but I have find nothing usefoul by now ... any guess? Tnx ! -- Best regards, Alessio mailto:[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] AGI and EXEC function CONFIRMATION
Clayton/group-- Thanks for the quick response... I think we may be on a different wavelength on the queue problem, though. I cant get the following native agi commands to Return ***BEFORE*** a user response-- which makes sense-- because these commands want to return with the response DATA. Stream File Get Data By the looks of your code fragment--- I think you are getting a return only ***AFTER*** a user presses a Key to generate DTMF or a timeout, yes? (please advise) If not-- Ill have another look at phpagi, which provides my interface to the AGI API. My particular problem lies in the fact that any Dialplan Application (ie: Background) launched through the native ***EXEC*** agi command seems to block until after the audio segment plays completely-- which prevents the queueing functionality provided by the Background application. Im still tinkering with this... but thought someone might have some more insight. Thanks again! Cheers- JJQ _ Check out Election 2004 for up-to-date election news, plus voter tools and more! http://special.msn.com/msn/election2004.armx ___ 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] Astricon hotel recommendations.....?
Rick, I just spoke with the hotel and I have special dispensation for AstriCon attendees to ride the Marriott shuttle regardless of where they are staying. The hotel considers all attendees guests and will be happy to provide a pickup. In order to expedite the pickup, please call the hotel when you arrive at Brookhaven. The number is posted on the AstriCon web page. Thanks, Steven Steven Sokol Owner/Manager Sokol Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web:http://www.sokol-associates.com ASK ME ABOUT AstriCon 2004! http://www.astricon.net/ -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Nick Barnes Sent: Thursday, August 26, 2004 10:00 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Astricon hotel recommendations.? Rick L. Wilson, Sr. wrote: I lived there for many years. There should be buses to Brookhaven from most anywhere in the area. Sorry, I probably didn't do a good job of explaining... The problem is that the Mariott is 4 miles from Brookhaven, there doesn't seem to be any public transport between the hotel and Brookhaven and the hotel shuttle is for hotel residents only. So... How do I get from Brookhaven to the Marriott? Nick Barnes ___ 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] ISDN Card Recommendation
The ISDN cards that are so popular with Asterisk have an S/T interface, which is fine where they're used -- Europe. In the US, we need a U interface. Also, I've been told that certain EuroISDN cards can't be used in the US because they use E1 signalling (which I don't know anything about, so I can't say if they're right or wrong...). Rob On Thu, 26 Aug 2004 10:09:55 -0500, Paterson, Mark [EMAIL PROTECTED] wrote: What's the deal w/ this setup in the US? I'm a little new to the Asterisk thing and my company has an unused ISDN BRI provisioned for both data and voice (NT1) Is this not possible? Rgs, Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Fugina Sent: Thursday, August 26, 2004 9:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ISDN Card Recommendation Good luck finding a Linux/Asterisk-compatible ISDN PCI card that's usable in the US. I've had my eye out for a while, and haven't had any luck. If you do find something, and find someone who will sell it to you, please share it with the list... Rob On Thu, 26 Aug 2004 08:26:24 -0500, Paterson, Mark [EMAIL PROTECTED] wrote: I'm running Asterisk 1.0 RC2 on a RedHat 9.0 box. I have a ISDN BRI line that I would like hook up to my Asterisk server and would like to ask the group what you guys would recommend as far as isdn cards that install easily into the Linux and asterisk environment. Rgs, Mark ___ 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 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] Error Compiling MySQL Friends
Hi, try issue: export lang=C before make. Regards, R. Wong - Original Message - From: imail [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, August 26, 2004 11:13 PM Subject: Re: [Asterisk-Users] Error Compiling MySQL Friends Thanks Flynn but still no go. I looked at line 141 and it seems to be fine. elifeq ($(USE_SIP_MYSQL_FRIENDS),1) I also tried removing the comma, and putting in a tab but I get the same error. I havent made any changes to the file, it was downloaded automatically via cvs. Any other thoughts? - Original Message - From: el Flynn [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, August 26, 2004 1:45 AM Subject: Re: [Asterisk-Users] Error Compiling MySQL Friends imail wrote: All, I edited the Makefile under asterisk/channels and set: USE_MYSQL_FRIENDS=1 USE_SIP_MYSQL_FRIENDS=1 When I do a make clean ; make install I get the following for x in res channels pbx apps codecs formats agi cdr astman stdtime; do make -C $x clean || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk/res' rm -f *.so *.o .depend make[1]: Leaving directory `/usr/src/asterisk/res' make[1]: Entering directory `/usr/src/asterisk/channels' Makefile:141: *** missing separator. Stop. make[1]: Leaving directory `/usr/src/asterisk/channels' make: *** [clean] Error 1 Could it be that your problem is coming from the error: Makefile:141: *** missing separator. Stop. From the rest of the output it doesn't seem to imply there's something missing where MySQL is concerned. Googling on makefile and missing separator gave me this link that may be of help: http://www.cygwin.com/ml/cygwin/2003-07/msg00341.html Although I may be way off, you could try that out first and see if it doesn't solve the problem. Cheers, Flynn ___ 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] Anyone using Asterisk on Slackware 9?
On Thursday 26 August 2004 11:22, Chris Blunt wrote: Can anyone give me a list of what packages I need to install as I don't want X an all the associated bloat? Slackware 9.1 here: # ls -l /var/log/packages/ | cut -c 57- Authen-SASL-2.07-i386-1 Crypt-DES-2.03-i386-1 Digest-HMAC-1.01-i386-1 Digest-MD5-2.33-i386-1 Digest-SHA1-2.10-i386-1 Frontier-RPC-0.07b4-i386-1 HTML-Parser-3.36-i386-1 HTML-Tagset-3.03-i386-1 Net-Jabber-1.30-i386-1 Net-SNMP-4.1.2-i386-1 RRDs-1.0.48-i386-1 URI-1.30-i386-1 XML-Parser-2.34-i386-1 XML-Stream-1.21-i386-1 aaa_base-9.1.0-noarch-1 aalib-1.4rc5-i386-1 acct-6.3.2-i386-1 acpid-1.0.2-i486-1 apmd-3.0.2-i386-1 asterisk-20040806-i386-1 asterisk-samples-20040806-i386-1 at-3.1.8-i386-1 bash-2.05b-i486-3 bc-1.06-i386-2 bin-8.5.0-i386-1 bind-9.2.2_P3-i486-1 bsd-games-2.13-i386-6 bzip2-1.0.2-i386-4 coreutils-5.0-i486-4 cpio-2.5-i386-1 curl-7.10.7-i486-1 cvs-1.11.6-i486-1 cxxlibs-5.1.0-i486-1 db1-1.85-i386-1 db2-2.4.14-i386-1 db3-3.3.11-i386-3 dcron-2.3.3-i386-4 devfsd-1.3.25-i386-2 devs-2.3.1-noarch-18 dhcp-3.0pl2-i386-1 dhcpcd-1.3.22pl4-i386-1 diffutils-2.8.1-i386-1 elflibs-9.1.0-i486-2 elvis-2.1_4-i386-1 etc-5.1-noarch-5 ethereal-0.10.5-i386-1 expat-1.95.6-i486-2 findutils-4.1.7-i386-1 flac-1.1.0-i386-1 gawk-3.1.3-i486-1 gdbm-1.8.0-i386-3 genpower-1.0.3-i486-1 gettext-0.11.5-i386-1 glib-1.2.10-i386-2 glibc-solibs-2.3.2-i486-1 glibc-zoneinfo-2.3.2-noarch-1 gmp-4.1.2-i486-2 gnet-2.0.4-i486-1 gnupg-1.2.3-i486-1 grep-2.5-i386-2 groff-1.17.2-i386-3 gzip-1.3.3-i386-2 hdparm-5.3-i386-1 hotplug-2003_08_05-noarch-3 i2c-2.8.4-i386-1 imlib-1.9.14-i486-2 infozip-5.50-i486-2 iptables-1.2.8-i486-1 iptraf-2.7.0-i386-1 kernel-modules-2.4.22-i486-2 less-381-i386-1 libgr-2.0.13-i386-2 libjpeg-6b-i386-4 libmad-0.15.0b-i486-1 libmng-1.0.5-i486-1 libpng-1.2.5-i386-1 libpri-20040629-i386-1 libtiff-v3.6.0-i386-3 libungif-4.1.0b1-i386-4 libusb-0.1.7-i386-1 libwww-perl-5.79-i386-1 libxml2-2.5.11-i486-2 libxslt-1.0.33-i486-1 lilo-22.5.7.2-i386-1 lm_sensors-2.8.6-i386-1 logrotate-3.6.8-i486-1 lsof-4.68-i486-1 lvm-1.0.7-i486-1 lynx-2.8.4-i386-5 m4-1.4-i386-2 make-3.80-i386-1 man-1.5l-i386-1 man-pages-1.60-noarch-1 mc-4.6.0-i386-1 minicom-2.00.0-i386-1 module-init-tools-0.9.14-i486-2 mpg321-0.2.10-i486-2 nc-1.10-i386-1 ncftp-3.1.6-i486-1 ncurses-5.3-i386-1 nfs-utils-1.0.6-i486-1 nss_ldap-217-i386-1 ntp-4.1.2-i486-2 oggutils-1.0-i386-3 openldap-client-2.2.8-i486-1rob openssh-3.7.1p2-i486-1 openssl-0.9.7b-i486-2 openssl-solibs-0.9.7b-i486-2 orbit-0.5.17-i386-1 pango-1.2.5-i486-1 pciutils-2.1.11-i386-4 pcre-4.4-i486-1 perl-5.8.0-i486-5 pkgconfig-0.15.0-i486-1 pkgtools-9.1.0-i486-4 popt-1.7-i386-1 portmap-5.0-i486-1 procps-2.0.16-i486-2 proftpd-1.2.8p-i486-1 python-2.3.1-i486-1 python-tools-2.3.1-noarch-1 quota-3.09-i486-1 raidtools-1.00.3-i386-1 readline-4.3-i486-3 reiserfsprogs-3.6.11-i486-1 rsync-2.5.6-i386-1 screen-3.9.15-i486-2 sed-3.02-i486-1 sgml-tools-1.0.9-i386-8 shadow-4.0.3-i486-8 slocate-2.7-i486-2 smartmontools-5.1_18-i486-1 sox-12.17.4-i486-2 spandsp-0.0.1-i386-3 strace-4.4.98-i486-2 sudo-1.6.6-i386-1 sysklogd-1.4.1-i486-8 sysvinit-2.84-i486-36 t1lib-1.3.1-i386-2 tar-1.13.25-i386-1 tcpdump-3.7.2-i386-1 tcpip-0.17-i486-24 traceroute-1.4a12-i386-2 usbutils-0.11-i386-1 utempter-0.5.2-i486-2 util-linux-2.12-i486-1 vim-6.2-i486-1 wget-1.8.2-i386-2 xfsprogs-2.5.6-i486-1 yptools-2.8-i486-3 zaptel-20040806-i386-1 zlib-1.1.4-i386-3 Obviously some of these aren't necessary but it gives you a good starting point. My slackware 9.1 install for asterisk is under 400MB, and if I got rid of the perl stuff it'd likely be under 350MB. Seriously though it's not hard to figure this out. ldd the shared libs and programs asterisk needs and then track back the dependencies if you're really serious about cutting it back. I consider 500M install pretty decent though, especially since it'll fit into a 512M CF card if you really wanted to, but I'd suggest mounting /var and /tmp off the card and seeing what else wants write access. :-) -A. ___ 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] Overhead Paging
- Original Message - From: Brian Pavane [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, August 26, 2004 5:12 AM Subject: [Asterisk-Users] Overhead Paging All, I am currently implementing a VoIP PBX, and need to deal with the paging situation. I would prefer to do paging via overhead speakers. My plan is to connect a Paging Unit to an FXS port of an IAD, and assign an extension to that port. I would then simply be able to call that extension, and have my call patched through to the overhead speakers. Has anyone implemented this type of setup, if so, what type of paging unit did you deploy, did you require an external amplifier or power supply, and how many speakers were you able to connect to the unit? As it stands, I will need between 4 and 8 speakers, and some of the speakers will be 400 feet from the main telco closet. Any thoughts, comments, and suggestions that you can shed on this topic would be much appreciated. If you have other methods of implementing overhead paging, I would also be interested. -Brian The Valcom units don't mention SIP... they must be some proprietary protocol, also says it's Windows based... eww... Anything that you can do to an FXS port on a regular PBX you should be able to do to the FXS port of an IAD (within reason of course). You don't usually connect the speakers directly to the FXS, usually there's a device like an inline power supply that provides voltage and audio to the speakers or bullhorn... The GrandStream HT486 has a nice intercom function which sends a BEEP! before answering. The SPA-2000 also has an intercom but doesn't provide the beep... -Chris ___ 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] IPDialog call transfer
Hi, Has anyone had any experience of doing supervised transfers with an IPDialog SipToneII SIP phone and asterisk? When The phone sends the REFER message asterisk errors with a message saying it canot find the call referenced by the Replaces Header. Thank you. Jonathan ___ 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] Overhead Paging
On Thu, 2004-08-26 at 10:19, Rich Adamson wrote: Any thoughts, comments, and suggestions that you can shed on this topic would be much appreciated. If you have other methods of implementing overhead paging, I would also be interested. One (of many) ways to accomplish it is simply based on using a Cisco 7940/7960 phone configured with paging, and pipe the audio to an amplifier input. If you're planning on deploying the Cisco phones already, then using that approach basically has built-in sparing covered. If the Cisco phone is already in use, and a page comes in, how should this be handled? It seems to show like an incoming call, and does not auto answer if you are on the phone already. Then it keeps ringing till you get off the phone. Or am I doing something wrong? This is the setup that I tried: http://www.voip-info.org/wiki-Cisco+7940-7960+auto-answer+config -- respectfully, Joseph === -= ** = ___ 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] ISDN Card Recommendation
Rob Fugina schrieb: The ISDN cards that are so popular with Asterisk have an S/T interface, which is fine where they're used -- Europe. In the US, we need a U interface. Also, I've been told that certain EuroISDN cards can't be used in the US because they use E1 signalling (which I don't ... Hi, is the AVM Fritz!Card PCI not compatible with US american ISDN? It's available for 55 EUR. Roger. ___ 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] RC2 and VoicePulse
I upgraded to RC2 over the weekend, and my IAX2 links to VoicePulse quit working. Nothing in the configs has changed. I confirmed the parameters with VoicePulse. I am able to make other IAX2 calls. Is anyone else having this problem? Thanks, Mike ___ 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] Overhead Paging
I have several Cisco 7940's laying around, how do I piple the speakerphone through external speakers? I understand the amplifier part, but how do you get RCA/2.5mm outputs from the Cisco? For now, we just configured a line 2 on all our phones with auto answer and, using the trick found in the wiki, when we page, every phone turns on and broadcasts whatever we say. However, there is an antiquated speaker system in the ceiling above that nobody knows anything about, it'd be neat to pipe a dedicated 7940 with Auto-Answer to the above Intercom. Rich Adamson wrote: I am currently implementing a VoIP PBX, and need to deal with the paging situation. I would prefer to do paging via overhead speakers. My plan is to connect a Paging Unit to an FXS port of an IAD, and assign an extension to that port. I would then simply be able to call that extension, and have my call patched through to the overhead speakers. Has anyone implemented this type of setup, if so, what type of paging unit did you deploy, did you require an external amplifier or power supply, and how many speakers were you able to connect to the unit? As it stands, I will need between 4 and 8 speakers, and some of the speakers will be 400 feet from the main telco closet. Any thoughts, comments, and suggestions that you can shed on this topic would be much appreciated. If you have other methods of implementing overhead paging, I would also be interested. If you search the archives I think you'll find this discussed several times. One (of many) ways to accomplish it is simply based on using a Cisco 7940/7960 phone configured with paging, and pipe the audio to an amplifier input. If you're planning on deploying the Cisco phones already, then using that approach basically has built-in sparing covered. ___ 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
