[Asterisk-Users] Diconnectiong after 15s when calling DIAX to DIAX (Tony?)
Hi, There is someone (Tony?) with disconnection problems (after about 15s) when calling between two DIAX phones? I have a voicemessage regarding this issue, without any contact address. If yes, please send me more details about configuration (iax.conf and extensions.conf files, IAX mode, etc.). As another DIAX user requested that, I'll put on my site some sample configurations files to be used with DIAX. If you leave me voice messages using CallMe function from DIAX, please do one of the following: - put your e-mail address as CallerID or - put an IAXTEL/FWD number as CallerID in order to be able to answer you. or leave your e-mail address in the voice message. Thank you and best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FARFON lives!
Some of you have been following our progress on http://farfon.convergence.com.pk as we blundered our way through the development of a low-cost ethernet IP phone that does IAX and augments the client options currently available for the kick-assterisk server. With help from the denizens of #asterisk and kind words of advice from Mr. Spencer and the rest of the gang ... we're proud to have accomplished our final milestone #10 yesterday with successful network boot of the FARFON. We've got 3 (mostly) functional, but not very pretty to look at prototypes now. Next step is to build 50 pre-production units which will be sent to those wise and sundry. So, if you're interested in a (possibly slight late) Christmas stocking stuffer for your favourite IAX-head read on. This invitation to express your interest in testing the preproduction unit is primarily meant for those who are: a) actively developing IAX and * and can give feedback b) thinking about deploying large * installations and need phones As a precursor to the order stage: a) you will be expected to sign an NDA and abide by it b) the price for the preproduciton units will be sub EUR100 c) you will be expected to pay 50% in advance, and 50% on shipping We expect to start manufacturing the 50 units this coming week, with first units shipping end-Dec. Plase mail me OFFLIST for further instructions. I'll shortly be putting up a commercial website which will have the PRODUCT BULLETIN, pictures of the plastic enclosure and other information. Thanks for being there for us, we'll do our best to reciprocrate. Hail IAXY! Hail ZapBRI! Hail *! -- Mirza Wasim Baig | Principal Consultant | Convergence, Islamabad Pakistan #48, St 32, Sector F-6/1, Islamabad, Pakistan 44000 | US: +1(800)460-1446 VOX: +92(51)282-0628 | FAX: +92(51)282-0621 | GSM: +92(300)850-8070 This mail is confidential intended solely for the use of the addressee and part of the world domination conspiracy. p.s. no, its not a hairdryer, farfon is just a code name ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RxFAX application
Hi all,
I have installed FAX app as described in several mails.
When a fax call is received, I get the following in the * console:
-- Starting simple switch on 'Zap/1-1'
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing Ringing(Zap/1-1, ) in new stack
-- Executing Wait(Zap/1-1, 2) in new stack
-- Executing DBget(Zap/1-1, CIDTMP=phonebook/0722123456) in new
stack
-- DBget: varname=CIDTMP, family=phonebook, key=0722123456
-- DBget: set variable CIDTMP to Dan-Mobile
-- Executing SetCallerID(Zap/1-1, 0722123456) in new stack
-- Executing SetCIDName(Zap/1-1, Dan-Mobile) in new stack
-- Executing SetLanguage(Zap/1-1, ro) in new stack
-- Executing Dial(Zap/1-1, SIP/101SIP/103IAX/dan|30|Ttr) in new
stack
-- Called 101
-- Called 103
-- SIP/101-65f1 is ringing
-- SIP/103-7dda is ringing
-- Redirecting Zap/1-1 to fax extension
== Spawn extension (inbound-analog, fax, 0) exited non-zero on 'Zap/1-1'
== Spawn extension (inbound-analog, fax, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
This is what I have in extensions.conf:
[inbound-analog]
exten = fax,1,RxFAX(/var/spool/asterisk/incoming/${CALLERIDNUM}.tif)
exten = fax,2,Hangup
exten = s,1,Answer
exten = s,2,Ringing
exten = s,3,Wait(2)
exten = s,4,DBget(CIDTMP=phonebook/${CALLERIDNUM})
exten = s,5,SetCallerID(${CALLERIDNUM})
exten = s,6,SetCIDNAME(${CIDTMP})
exten = s,7,SetLanguage(ro)
exten = s,8,Dial(SIP/101SIP/103IAX2/dan,30,Ttr)
exten = s,9,Wait(1)
exten = s,10,Voicemail2(u101)
exten = s,9,Hangup
exten = s,109,Wait(3)
exten = s,110,Goto(8)
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Re: [Asterisk-Users] RxFAX application
Hi, I have started the * server in console mode (-vc) and this is what I get now (no file saved and disconnected): *CLI -- Starting simple switch on 'Zap/1-1' -- Executing Answer(Zap/1-1, ) in new stack -- Executing Ringing(Zap/1-1, ) in new stack -- Executing Wait(Zap/1-1, 2) in new stack -- Executing DBget(Zap/1-1, CIDTMP=phonebook/0722123456) in new stack -- DBget: varname=CIDTMP, family=phonebook, key=0722123456 -- DBget: set variable CIDTMP to Dan-Connex -- Executing SetCallerID(Zap/1-1, 0722123456) in new stack -- Executing SetCIDName(Zap/1-1, Dan-Mobile) in new stack -- Executing SetLanguage(Zap/1-1, ro) in new stack -- Executing Dial(Zap/1-1, SIP/101SIP/103IAX/dan|30|Ttr) in new stack -- Called 101 -- Called 103 -- SIP/101-f828 is ringing -- SIP/103-3b8b is ringing -- Redirecting Zap/1-1 to fax extension == Spawn extension (inbound-analog, fax, 0) exited non-zero on 'Zap/1-1' -- Executing RxFAX(Zap/1-1, /var/spool/asterisk/incoming/0722285952.tif) in new stack Changed from phase 0 to 1 Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 38 37 36 35 34 33 32 31 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Store and forward Internet fax: no Real-time Internet fax: no Preferred octets: 256 Can receive fax Data signalling rate: 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 Inch-based resolution preferred: no Metric-based resolution preferred: no Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 c6 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 38 37 36 35 34 33 32 31 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Store and forward Internet fax: no Real-time Internet fax: no Preferred octets: 256 Can receive fax Data signalling rate: 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 Inch-based resolution preferred: no Metric-based resolution preferred: no Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 c6 f0 80 80 01 T2 timeout Start receiving document Sending ident CSI: 40 38 37 36 35 34 33 32 31 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Store and forward Internet fax: no Real-time Internet fax: no Preferred octets: 256 Can receive fax Data signalling rate: 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 Inch-based resolution preferred: no Metric-based resolution preferred: no Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 c6 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 38 37 36 35 34 33 32 31 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Store and forward Internet fax: no Real-time Internet fax: no Preferred octets: 256 Can receive fax Data signalling rate: 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 Inch-based resolution preferred: no Metric-based resolution preferred: no Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 c6 f0 80 80 01 T2 timeout Start receiving document Sending ident CSI: 40 38 37 36 35 34 33 32 31 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Store and forward Internet fax: no Real-time Internet fax: no Preferred octets: 256 Can receive fax Data signalling rate: 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 Inch-based resolution preferred: no Metric-based resolution preferred: no Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 c6 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 38 37 36 35 34 33 32 31 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Store and forward Internet fax: no Real-time Internet fax: no Preferred octets: 256 Can receive fax Data signalling rate: 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 Inch-based resolution preferred: no Metric-based resolution preferred: no Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 c6 f0 80 80 01 T4 timeout in state 9 Sending ident CSI: 40 38 37
Re: [Asterisk-Users] RxFAX application
Hi Dan, Dan wrote: Hi, I have started the * server in console mode (-vc) and this is what I get now (no file saved and disconnected): [] It seems the software FAX modem is sending out its messages regularly, but never hears anything recognisable come back from the far end FAX machine. That could mean nothing comes back, or something distorted (maybe a wrong codec) comes back. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Off topic: suggestion for call center ?
Hi guys, I am in the process of establishing a Call Center. I need some suggestions from those who have already worked on such setups using Asterisk. My scenerio is: US T1 --- Asterisk gw 1 -[gsm compression]-- Asterisk gw 2 [with TDM10B] --- Sip Phones [Xlite] Calls from US are landing perfectly on the Xlite and tdm10b, and vice versa. However I am having some trouble as mentioned below: 1. Call quality on Tdm10b is good, but sometimes on Xlite, sound becomes choppy (like the voicemissed for few mili secs). I am using jitterbuffer=no is IAX.conf (if set to yes, sounds becomes more choppy). 2. There is lot of echo during the conferencing [on Xlite], when in a room more than 4 people talk at the same time. 3. I am using GSM compression on the Asterisk, and my Xlite also uses GSM, but when I switch from GSM to iLBC in the server (Xlite still using GSM), sound qulaity degrades. Plz recommend your suggestions, and is Xlite good for Call Center ?? Further the echo that I am getting, can be caused due to normal headphones ??? can some guide me which headphone is best in terms of echo cancelling and noise reductions. Did anyone tried the Plantronics PLA-H161N (http://www.thephonesource.com/PLA-H161N.htm) ?? TIA Azher Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing
Re: [Asterisk-Users] RxFAX application
Hi Steve, - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, December 07, 2003 10:36 AM Subject: Re: [Asterisk-Users] RxFAX application Hi Dan, Dan wrote: Hi, I have started the * server in console mode (-vc) and this is what I get now (no file saved and disconnected): [] It seems the software FAX modem is sending out its messages regularly, but never hears anything recognisable come back from the far end FAX machine. That could mean nothing comes back, or something distorted (maybe a wrong codec) comes back. The fax is sent from a iPaq H3870 PocketPC through Bluetooth and GSM (Ericsson T68i) using KSE TrueFax application. I have tested before to see if it can be received on a standard FAX machine and it work perfect. What Fax Class is supported by your app? Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFAX application
Dan wrote: Hi Steve, - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, December 07, 2003 10:36 AM Subject: Re: [Asterisk-Users] RxFAX application Hi Dan, Dan wrote: Hi, I have started the * server in console mode (-vc) and this is what I get now (no file saved and disconnected): [] It seems the software FAX modem is sending out its messages regularly, but never hears anything recognisable come back from the far end FAX machine. That could mean nothing comes back, or something distorted (maybe a wrong codec) comes back. The fax is sent from a iPaq H3870 PocketPC through Bluetooth and GSM (Ericsson T68i) using KSE TrueFax application. I have tested before to see if it can be received on a standard FAX machine and it work perfect. What Fax Class is supported by your app? I'm not sure what you mean by Class. That term is usually applied to FAX modem interfaces. It supports V.29 at 9600 and 7200bps. The version I am testing now also supports V.27ter at 4800 and 2400. However, you are not getting that far. rxfax is not even seeing the initial exchange using 300bps/V.21. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFAX application
Hi, - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, December 07, 2003 11:11 AM Subject: Re: [Asterisk-Users] RxFAX application It seems the software FAX modem is sending out its messages regularly, but never hears anything recognisable come back from the far end FAX machine. That could mean nothing comes back, or something distorted (maybe a wrong codec) comes back. The fax is sent from a iPaq H3870 PocketPC through Bluetooth and GSM (Ericsson T68i) using KSE TrueFax application. I have tested before to see if it can be received on a standard FAX machine and it work perfect. What Fax Class is supported by your app? I'm not sure what you mean by Class. That term is usually applied to FAX modem interfaces. It supports V.29 at 9600 and 7200bps. The version I am testing now also supports V.27ter at 4800 and 2400. However, you are not getting that far. rxfax is not even seeing the initial exchange using 300bps/V.21. There is anything I can do to make it work? Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Project Critique
On Sat, 2003-12-06 at 17:55, Cameron Jacobson wrote: I have just started laying out the plans for my first project using Asterisk. I am very interested at this stage in getting much needed feedback, critiquing my approach. What are the ups and downs going to be if I develop this project as follows: -The client wants to connect some phone reps in India through a VoIP to their clients. -There will be 3 phone lines, and 1 broadband internet connection. -Since there will be very little bandwidth requirements (unless they scale), I figure just let the workers in India make and receive calls using NetMeeting, or GnomeMeeting, or some other H323, or SIP compatible client, directly connecting to the Asterisk box in the U.S. -These clients will connect, via the internet, to an Asterisk box in their office in the U.S. That box (regardless if it uses Digium or Quicknet) will negotiate all 3 phone connections with their respective H323 or SIP connections. Another question is: If the workers are in an office-space, and in order to prevent the need for an additional asterisk PBX in India, is it easy to run each of the workers' connections on separate ports? (ie. port 5060 for phone # 555-1212, port 5061 for 555-1213, and port 5062 for 555-1214 ? This way each NetMeeting, or GnomeMeeting connection coming from India can simply run behind a NAT router, instead of setting up a separate Asterisk PBX. What kind of connection is you client in India getting? What is the sustained bandwidth you can achieve to the US(or wherever you are originating the calls)? How reliable is the connection, and have you made fail over plans? While GSM won't provide the number of connections that G729 will on a given network connection, you can expand at anytime without fear of loosing your investment. To get GSM to India and keep quality up, you will want to deploy a asterisk machine behind the NAT in India. Then if you wanted to use SIP or H323, it will be much less of a problem as it would be on the same side of the NAT. It would also be possible to use a TDM400 card and use real telephones so as not to loose your phone connections when the windows based computers fail. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vonage sending Motorola gear now?
Its the VT1000 http://broadband.motorola.com/catalog/productdetail.asp?ProductID=212 We have looked everywhere for it but looks like no distributor sells it right now. - Original Message - From: Brian Capouch [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, December 07, 2003 4:50 AM Subject: [Asterisk-Users] Vonage sending Motorola gear now? I got a call from an ISP friend tonight who said he is getting calls from people who are getting signed up with Vonage. Instead of sending them ATA186s, apparently they're receiving something made by Motorola. They apparently work significantly differently than the Cisco units, and there have been some problems. Anybody know anything further? Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vonage sending Motorola gear now?
--On Sunday, December 07, 2003 09:36:14 -0500 TeleSIP [EMAIL PROTECTED] wrote: Its the VT1000 http://broadband.motorola.com/catalog/productdetail.asp?ProductID=212 We have looked everywhere for it but looks like no distributor sells it right now. Maybe because it's a new variant of the VT1000. PacketCable doesn't use SIP (it uses a derivative of MGCP) so the product may not yet be shipping with the SIP code. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CVS UPDATE Busted !!!!!
hi, i am getting make update error in asterisk directory = it is possible that you first compiled asterisk and unrecognized request ' then zaptel thus' cvs update: dying gasp from cvs.digium.com unexpected = Any clues? Thanks a lot in advance. Ricky ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some success with linux 2.6 and wcfxo
I for one would be very interested in seeing your moded makefiles, I'm also trying to use 2.6 wherever possible. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some success with linux 2.6 and wcfxo
same here :) On Sunday 07 of December 2003 16:23, Dave Cotton wrote: I for one would be very interested in seeing your moded makefiles, I'm also trying to use 2.6 wherever possible. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Maint.
What kind of stability / reliability are people currently experiencing with the Linux / Asterisk combination? We will be running 3-10 SIP phones from India to US using nothing more than regular cable / dsl connections from both locations. People have had months of uptime. I would be more concerned with the reliability of your DSL/Cable reliability. You should also use a Bandwidth calculator (like http://www.packetizer.com/iptel/bandcalc.html) to figure out how much bandwidth you're going to need and compare that to how much will be available. Also, what make / model SIP phone do you recommended that would allow us to configure the phones to work on alternate ports (or is this a standard configuration option on most SIP phones) ? Most do. I would reccomend Grandstream phones for your application, because they're cheap, easy, and tested with Asterisk. Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hybrid T1 Service (WAS: Channelbank Recomendation and GS102 question)
-Original Message- From: Walker Haddock Sent: Thursday, December 04, 2003 7:54 PM To: [EMAIL PROTECTED] We have an installation with 9 inbound voice channels (one is the fax) and 768K data. It is a Hybrid PRI. It terminates into a T100P. It is working great! The cost was better than the POTS plus data. This is a service that I'm interested in selling. Would you be willing to share with me (the list) exactly how you have this set up (read: your configuration files)? I've never used linux as a router, and am a bit leary of doing this and selling it as a supported service. I've got the voice stuff down I think, my primary interest is in how you accomplished the data portion. Thanks, -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Project Critique
Philipp von Klitzing wrote: For H323 you'll need to install a gatekeeper next to Asterisk and fiddle with h323 or oh323 (I love to live dangerously, hit me Jeremy). :- Moreover NetMeeting doesn't work through NAT. A gatekeeper is TOTALLY optional in an H.323 network. Read the spec. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channelbank Recomendation and GS102 question
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Thursday, December 04, 2003 8:48 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Channelbank Recomendation and GS102 question At 8:15 PM -0500 12/4/03, Jim Flagg wrote: - Original Message - From: Walker Haddock [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, December 04, 2003 7:54 PM Subject: Re: [Asterisk-Users] Channelbank Recomendation and GS102 question We have an installation with 9 inbound voice channels (one is the fax) and 768K data. It is a Hybrid PRI. It terminates into a T100P. It is working great! The cost was better than the POTS plus data. Can I ask what Telephone/Internet service provider you are getting this from? Does anybody else have a setup like this? Very interesting. I've had now two fights with providers (Verizon and SBC) who would not offer such a service, claiming that it was impossible to hybridize a PRI. I think that's a great offering, and of course, it is possible, and especially appealing for Asterisk users. I, too, would be interested in hearing from what vendor you are getting such a service. John, Check the front of your local phonebook for CLEC listings. In your area, I'd expect to find at a bunch listed, and at least two or three that are facilities based, capable of serving most areas in the Willamette Valley (Vancouver down to Eugene). If not, perhaps there's a good business for you to investigate. =D Our CLEC here, KMC Telecom, does the hybrid T1 thing as a matter of course. I can have a 6x6 system delivered to my customers for less than $400/month (.09/local connect), or unlimited local outbound for less than $500/month. KMC even provides the customer with a Lucent Connectreach, which breaks out the POTS lines and can hand off the data either as a FT1 or Ethernet. I'd like to play with using * to do it all, but need to find a qualified guinea pig first. -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some success with linux 2.6 and wcfxo
oops, my apologiee for attaching the entire source of the drivers :( Here's a
diff
Index: wcfxo.c
===
RCS file: /usr/cvsroot/zaptel/wcfxo.c,v
retrieving revision 1.21
diff -a -u -r1.21 wcfxo.c
--- wcfxo.c 17 Nov 2003 22:09:40 - 1.21
+++ wcfxo.c 7 Dec 2003 17:04:12 -
@@ -387,7 +387,7 @@
static void wcfxo_stop_dma(struct wcfxo *wc);
static void wcfxo_restart_dma(struct wcfxo *wc);
-static void wcfxo_interrupt(int irq, void *dev_id, struct pt_regs *regs)
+irqreturn_t wcfxo_interrupt(int irq, void *dev_id, struct pt_regs *regs)
{
struct wcfxo *wc = dev_id;
unsigned char ints;
@@ -402,7 +402,7 @@
if (!ints)
- return;
+ return IRQ_NONE;
if (ints 0x0c) { /* if there is a rx interrupt pending */
#ifdef ENABLE_TASKLETS
@@ -425,12 +425,12 @@
printk(FXO PCI Master abort\n);
/* Stop DMA andlet the watchdog start it again */
wcfxo_stop_dma(wc);
- return;
+ return IRQ_HANDLED;
}
if (ints 0x20) {
printk(PCI Target abort\n);
- return;
+ return IRQ_HANDLED;
}
if (1 /* !(wc-report % 0xf) */) {
/* Check for BATTERY from register and debounce for 8 ms */
@@ -504,6 +504,7 @@
#endif
}
+ return IRQ_HANDLED;
}
static int wcfxo_setreg(struct wcfxo *wc, unsigned char reg, unsigned char value)
Index: wcfxs.c
===
RCS file: /usr/cvsroot/zaptel/wcfxs.c,v
retrieving revision 1.39
diff -a -u -r1.39 wcfxs.c
--- wcfxs.c 23 Nov 2003 23:09:46 - 1.39
+++ wcfxs.c 7 Dec 2003 17:04:14 -
@@ -471,7 +471,7 @@
return 0;
}
-static void wcfxs_interrupt(int irq, void *dev_id, struct pt_regs *regs)
+irqreturn_t wcfxs_interrupt(int irq, void *dev_id, struct pt_regs *regs)
{
struct wcfxs *wc = dev_id;
unsigned char ints;
@@ -487,12 +487,12 @@
/* Stop DMA, wait for watchdog */
printk(FXS PCI Master abort\n);
wcfxs_stop_dma(wc);
- return;
+ return IRQ_HANDLED;
}
if (ints 0x20) {
printk(PCI Target abort\n);
- return;
+ return IRQ_NONE;
}
for (x=0;x4;x++) {
@@ -535,7 +535,7 @@
wcfxs_transmitprep(wc, ints);
}
-
+ return IRQ_HANDLED;
}
static int wcfxs_proslic_insane(struct wcfxs *wc, int card)
@@ -1277,6 +1277,7 @@
outb(0x3f, wc-ioaddr + WC_MASK0);
/* No external interrupts */
outb(0x00, wc-ioaddr + WC_MASK1);
+
}
static void wcfxs_restart_dma(struct wcfxs *wc)
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Re: [Asterisk-Users] some success with linux 2.6 and wcfxo
FYI, the usual place for patch's is on bugs.digium.com and then a little link in an email here to let ppl know about it .. This allows a single place where these get reviewed for inclusion in cvs ..nice work ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] console sound
On Sat, 06 Dec 2003 10:10:21 -0700, Michael Welter wrote Hi, I have a RH9 system with an onboard VIA sound chip. According to the archives, VIA won't work for asterisk. So, I disabled the VIA and I purchased a Creative Labs Soundblaster PCI 128-Voice soundcard ($13). This card is on the approved RedHat list. However, the documentation inside the package says VIBRA 128. Anyhow, kudzu doesn't see then card. The soundcard detection program says Ensoniq, ES1371 (AudioPCI-97), module es1371, which seems normal. However, there is no sound. Does anyone have experience with this? Thanks, Mike Hi My SB128PCI CT5880 (don't know what's the name on the box, got a raw card w/o anything) half-works on OSS drivers (es1371). I cannot record audio, playback works OK. Alsa (http://www.alsa-project.org/) works fine from command line and from my programs, gnophone bitches about No input space. After removing the check, works great. Disclaimer: didn't try with chan_oss BTW, with ALSA, you might be even able to use the VIA? my 0.02PLN Grzegorz Nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some success with linux 2.6 and wcfxo
It's not really intended to be applied at this stage since it will break 2.4. I am intending to update the bug in digiums database with my findings once I get the chance. On Sun, Dec 07, 2003 at 09:59:16AM -0800, TC wrote: FYI, the usual place for patch's is on bugs.digium.com and then a little link in an email here to let ppl know about it .. This allows a single place where these get reviewed for inclusion in cvs ..nice work ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Tristan 'Minty' Colgate [EMAIL PROTECTED] | ICQ #154577755 --- I don't mean to sound bitter, cold, or cruel, but I am, so that's how it comes out - Bill Hicks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] console sound
Thanks guys, I found a SoundBlaster 16 PCI at a different CompUSA store, and everything is working perfect. Grzegorz Nosek wrote: On Sat, 06 Dec 2003 10:10:21 -0700, Michael Welter wrote Hi, I have a RH9 system with an onboard VIA sound chip. According to the archives, VIA won't work for asterisk. So, I disabled the VIA and I purchased a Creative Labs Soundblaster PCI 128-Voice soundcard ($13). This card is on the approved RedHat list. However, the documentation inside the package says VIBRA 128. Anyhow, kudzu doesn't see then card. The soundcard detection program says Ensoniq, ES1371 (AudioPCI-97), module es1371, which seems normal. However, there is no sound. Does anyone have experience with this? Thanks, Mike Hi My SB128PCI CT5880 (don't know what's the name on the box, got a raw card w/o anything) half-works on OSS drivers (es1371). I cannot record audio, playback works OK. Alsa (http://www.alsa-project.org/) works fine from command line and from my programs, gnophone bitches about No input space. After removing the check, works great. Disclaimer: didn't try with chan_oss BTW, with ALSA, you might be even able to use the VIA? my 0.02PLN Grzegorz Nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IaxTel seems down
Would have probably been more appropriate to at least announce that iax was going to disappear at some specific date, as opposed to folks randomly discoverying it and chasing problems. (Kind of related to why there isn't a marketing plan.) Sorry, it was something of a side effect of some analysis we were doing on the bug that was causing iaxtel to crash periodically (now fixed). I let someone talk me into nobody uses it anyway, don't worry about it. My apologies Good, glad to hear things are better. Without getting into too much techie detail, what was the root problem? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
IAXTel Info (was RE: [Asterisk-Users] IaxTel seems down)
Good, glad to hear things are better. Without getting into too much techie detail, what was the root problem? There was just race that I introduced a while back. If calls came in while a reload was taking place in IAX2, bad things would happen. Now it's fixed. Originally I was thinking it had to do with SIP to IAX2 conversion but this ended up not being the case. However, in order to make iaxtel as scalable as possible, it still would be best to have iaxtel strictly act as an IAX2 soft switch. We're also working to improve the ability to have iaxtel know about other VoIP providers so that we can provide a single peering exchange for everyone within flat E.164 address space. This work is sponsored greatly by VoicePulse who has donated equipment to the project. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: IAXTel Info (was RE: [Asterisk-Users] IaxTel seems down)
On Sunday 07 of December 2003 21:14, Mark Spencer wrote: Good, glad to hear things are better. Without getting into too much techie detail, what was the root problem? There was just race that I introduced a while back. If calls came in while a reload was taking place in IAX2, bad things would happen. Now it's fixed. Originally I was thinking it had to do with SIP to IAX2 conversion but this ended up not being the case. However, in order to make iaxtel as scalable as possible, it still would be best to have iaxtel strictly act as an IAX2 soft switch. We're also working to improve the ability to have iaxtel know about other VoIP providers so that we can provide a single peering exchange for everyone within flat E.164 address space. This work is sponsored greatly by VoicePulse who has donated equipment to the project. Mark Hmm we are in the meantime quite some iax providers who serve A-Z, meaning worldwide. How do you want to handle that ? cheers Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FARFON lives!
Are you guys using power over Ethernet? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, December 07, 2003 2:32 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] FARFON lives! Some of you have been following our progress on http://farfon.convergence.com.pk as we blundered our way through the development of a low-cost ethernet IP phone that does IAX and augments the client options currently available for the kick-assterisk server. With help from the denizens of #asterisk and kind words of advice from Mr. Spencer and the rest of the gang ... we're proud to have accomplished our final milestone #10 yesterday with successful network boot of the FARFON. We've got 3 (mostly) functional, but not very pretty to look at prototypes now. Next step is to build 50 pre-production units which will be sent to those wise and sundry. So, if you're interested in a (possibly slight late) Christmas stocking stuffer for your favourite IAX-head read on. This invitation to express your interest in testing the preproduction unit is primarily meant for those who are: a) actively developing IAX and * and can give feedback b) thinking about deploying large * installations and need phones As a precursor to the order stage: a) you will be expected to sign an NDA and abide by it b) the price for the preproduciton units will be sub EUR100 c) you will be expected to pay 50% in advance, and 50% on shipping We expect to start manufacturing the 50 units this coming week, with first units shipping end-Dec. Plase mail me OFFLIST for further instructions. I'll shortly be putting up a commercial website which will have the PRODUCT BULLETIN, pictures of the plastic enclosure and other information. Thanks for being there for us, we'll do our best to reciprocrate. Hail IAXY! Hail ZapBRI! Hail *! -- Mirza Wasim Baig | Principal Consultant | Convergence, Islamabad Pakistan #48, St 32, Sector F-6/1, Islamabad, Pakistan 44000 | US: +1(800)460-1446 VOX: +92(51)282-0628 | FAX: +92(51)282-0621 | GSM: +92(300)850-8070 This mail is confidential intended solely for the use of the addressee and part of the world domination conspiracy. p.s. no, its not a hairdryer, farfon is just a code name ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind NAT How to do it.
On Wed, 2003-12-03 at 15:34, William Waites wrote: localnet= internal ip of * machine? localnet should be the internal network address not the internal ip address. i.e. if your asterisk server is 192.168.0.245, localnet should be 192.168.0.0 Agreed, I was wrong before :) -- Leif Madsen [EMAIL PROTECTED] http://www.hacklocalhost.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call does not terminate correctly
We are using an MGCP configuration. There seems to be some incompatibilities between our Netergy T2 VOIP chip and Asterisk. This is how our Vendor sees it: Here's what I see.1. The first call is initiated. (CRCX) The interesting thing here is that the CA (Call Agent) tells us to go directly into sendrecv mode which means that we start streaming audio immediately. All other CAs that we've worked with do not instruct us to go to sendrecv mode until the number has been completely dialed.2. The call is terminated whenhung up. The call agent responds to this, but it never tells us to delete the connection and we continue to stream audio.3. The next call is attempted. We are now, not in the state that the call agent thinks we should be in and we are streaming audio to a UDP port that is now closed since the CA tore down the first call.4. The unit is rebooted. (The T2 is hard reset) The RSIP that is sent to the call agent basically resets the state machine and now the T2 and CA are in sync. I'm not sure why this is happening, but maybe Asterisk can help. It's clearly something in their code, but I can't really tell any more than that. Our sequence of events: 1) Made first phone call to cell phone. Call was successful left it on for a few minutes. Tried punching all kinds of digits while on the call. Hung up.2) Made second call. Picked up handset, was receiving dial tone. Tried first digit and received the error (buzzing sound from the handset) . The digit tone goes haywire and repeats itself over and over again (I think this is what creates the buzzing tone).Tried to make call while this was taking place. Hung up. 3) Reset T2.4) Made three-four more additional calls all worked after resetting T2. Any input would be greatly appreciated. Thanks, Jeff
RE: [Asterisk-Users] iax vs iax2 question
I'm trying to find this 'posting'. For some reason I'm missing it. Can anyone point it out please? - the host= setting (plus deny=/permit=) in particular is what can create the unexpected headaches if used with type=friend (some weeks ago there was an excellent posting on this issue, probably by J Todd), e.g.: you want to fix a host for the peer, but you might not want to fix a host for the user. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channelbank Recomendation and GS102 question
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: Thursday, December 04, 2003 3:48 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Channelbank Recomendation and GS102 question Very interesting. I've had now two fights with providers (Verizon and SBC) who would not offer such a service, claiming that it was impossible to hybridize a PRI. I think that's a great offering, and of course, it is possible, and especially appealing for Asterisk users. I, too, would be interested in hearing from what vendor you are getting such a service. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Unless the world o' telecom has changed there really isn't such a thing as a Hybrid PRI. You can order a PRI with either voice only, data only or voice+data. I used to order them all the time from GTE (now Verzion) when I ran an ISP here in Hawaii a few years back. We ordered voice+data to support was called data over voice. The only difference between an ISDN voice or data call is what is sent over the d channel during call setup (voice/data and 56k/64k). GTE used to charge by the minute for local isdn data calls but isdn voice calls were free. The Livingston Portmasters we used as access servers would answer the inbound call as a voice call and then look at the inbound data stream. If it was ppp the Portmaster would just process it as ppp data. It was a way of getting around gte's per minute data call charges. As far as GTE's switches (and CDR records) were concerned it was just a voice call. The person who you talked to at Verizon probably had no idea what hybrid was so they just said as usual that it didn't exist. Hell, most Telco sales I've experienced still don't even know that ISDN exists :-) Try ordering a data PRI and add voice or voice PRI and add data, either will work. I'll bet they are two different tariffs :-) To get 768k of data out of a pri you would need to bond 12 channels via multilink-ppp (rfc1618 as I recall). The linux 2.4 kernel supports multilink-ppp (mlpppd) over a number of different phy layers. I'm new * and just got my first * up and running (*very* cool so far, needs doc writers though :-). I'm expecting my GS phones Monday from pulver, I've never seen a T100P. I assume you setup the T100P to use X number of channels for voice and the remaining go to mlpppd for data. I'm guessing here but I assume you would have a hunt group assigned to the voice pri channels for inbound voice calls. If you wanted inbound data you would need a second hunt group assigned to the data channels. Is that a good guess Walker? :-) John Breeden Plum Hall, Inc. Hawaii ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming IAX2 problems with NuFone
I've been using NuFone with Asterisk for a while, but I've started seeing this error with incoming calls: NOTICE[114696]: File chan_iax2.c, Line 4581 (socket_read): Rejected connect attempt from 216.234.116.189, requested/capability 0x4/0x4 incompatible with our capability 0xff03. Outgoing works just fine, but I can't get incoming to work at all. Any ideas? I googled for the error, but I couldn't find anything. David -- David Coulsonemail: [EMAIL PROTECTED] Linux Developer / web: http://davidcoulson.net/ Network Engineer phone: (216) 533-6967 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Roaming Users
I'm trying to come up with an elegant solution to handle roaming users in a branch office scenario. I have a number of possible scenarios, none of which seem to completely solve the problem. Perhaps someone with a better feel of the interactions can help me out. Is the 'switch' statement useful in some way? What are the ins and outs of the 'switch' statement? Come to think of it, there was a post a while back regarding how * searches the dial plan. Does any one have a handy link to that message Any way, my example: a company has a number of branch offices. A few implementation scenarios include: a) One central hosted * against which all phones register. Phones requiring TFTP loads will need a local server or some sort of VPN'd TFTP connection to the hosted server. The canreinvite parameter on SIP phones can get complicated if we want to keep local calls local to the branch, but yet all off-net calls go to the hosted PBX for transmission to a gateway. This scenario is conducive to a roaming extension that can go from office to office. b) One central hosted * which handles DID routing and corporate Auto Attendant functions. Each branch office has a * for local extensions and voicemail. Each office will have to have a range of extensions assigned to it. This limits portability but effectively makes use of * trunking capability to get to the hosted * for off-net calls. There must be a way of merging the two scenarios so I can get: a) roaming user capability (user can take own extension [logically or physically] between offices) b) local * server at each office to handle trunking to hosted * c) central auto attendant for all offices D) local extension to extension calls stay local (don't cross the WAN to the hosted *) Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Phone Unprovisioned Message in IP 7940 ?
Hello all, I am newbie to Telephony world (IP and PSTN). Please excuse me if you find my questions very dumb. I am trying to configure my IP 7940 with the Asterisk, when phone boots up it only shows the message Phone Unprovisioned on the LCD panel. Under Settings--SIP Configuration--Line 1 Settings I noticed that Proxy Address is set the UNPROVISIONED, I am not sure why it is showing that though I did set proxy1_address: `129.82.44.223 in SIPDefault.conf, which is my Astersik server. Following SIP image is installed on the IP 7940. Application Load ID POS30203 My sip.conf has following lines added for the the Phone [810] type=friend secret=pass host=dynamic callerid=JOSE 810 defaultip=129.82.44.205 In my SIPmac.conf file I have made following entries # Line 1 appearance line1_name: 810 # Line 1 Registration Authentication line1_authname: 810 # Line 1 Registration Password line1_password: pass Do you see any problem here, Please let me know if I should give any more information. Regards Tony
Re: [Asterisk-Users] voip-info.org is a great Resource ..BUT
On Fri, 2003-12-05 at 01:11, Jonathan Tew wrote: Being farely new to the Asterisk scene and searching for documentation I was wondering why the asterisk.org site didn't run a wiki. There isn't anything as good as a wiki for collecting collaborative documentation. Over time someone might want to convert the knowledge contained in the wiki to more updated formal documentation, but a wiki sure would be a great thing to have. We're working on it :) IRC: #asterisk-doc -- Leif Madsen [EMAIL PROTECTED] http://www.hacklocalhost.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX 0.9.6 now available- some fixes included
- Original Message - From: Dan [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Friday, December 05, 2003 4:59 AM Subject: [Asterisk-Users] DIAX 0.9.6 now available- some fixes included Hi all, A new version (0.9.6) of DIAX is available for download at: http://www.laser.com/dante or http://www.geocities.com/tdanro There are no new functions, but some bugs fixed: What's new in version 0.9.6: - add Default_user locales as new language. The program language can be automatically selected based on default user locales on your system. You still can manually select the language, independent of the system locales. BUGS solved: -Windows height to small when Windows XP desktop theme is used; - if you select the language for the current locales in your system, English is used; - incoming call CallerID is not displayed correctly when IAX(1) is used; - cannot delete the second item from the phonebook memory. As usual, please send me any comments you may have or features requests. Best regards, Dan DIAX 0.9.6 gave me a little trouble tonight on WinME. (My wife's laptop, I don't use ME, honest!) I'll have to fiddle with it some more to get a list of exactly what's up, but this is what I remember: It started up in what I think was German. I went to what I guessed to be language and chose English. The action completed, but the language didn't change. I tried picking a few different languages from the list, then tried Ctrl E, and Ctrl S (for English, and Spanish). Somewhere in my trying to read the menus it crashed. It crashed on me once more before I gave up on trying to change the language. I walked through the menus and set up the username/password and dialed into my asterisk. I got into meetme, so it appears to be functioning, just stuck in another country... You hate me, don't you? I'll get the crash info next time, and I'll pick up a German-to-English dictionary just in case... - Andrew Thompson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Prefix the * character
Any ideas on how to do this one? FWD requires an * on certain calls as a prefix character, but I cannot seem to be able to get Prefix(*) to add that to the front of the extension that is dialed... Setting up an extension that dials (SIP/[EMAIL PROTECTED]) works just fine, but in trying to add the prefix to a dialed number, it acts as if the Prefix command was not even there, ie: exten = 1,1,Prefix(*) exten = *1,2,Dial(SIP/[EMAIL PROTECTED]) would dial [EMAIL PROTECTED], and not [EMAIL PROTECTED] Any ideas? Tnx Kris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prefix the * character
On Sunday 07 December 2003 22:48, Kris Stark wrote:
Any ideas on how to do this one?
FWD requires an * on certain calls as a prefix character, but I
cannot seem to be able to get Prefix(*) to add that to the front of
the extension that is dialed... Setting up an extension that dials
(SIP/[EMAIL PROTECTED]) works just fine, but in trying to add the
prefix to a dialed number, it acts as if the Prefix command was not
even there, ie:
exten = 1,1,Prefix(*)
exten = *1,2,Dial(SIP/[EMAIL PROTECTED])
would dial [EMAIL PROTECTED], and not [EMAIL PROTECTED]
Any ideas?
Prefix is an older application which was more useful prior to being
able to manipulate variables (the days of BYEXTENSION instead of
${EXTEN}). Instead, do:
Dial(SIP/[EMAIL PROTECTED])
-Tilghman
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Re: [Asterisk-Users] Prefix the * character
exten = _7X,2,Dial(SIP/*${EXTEN:[EMAIL PROTECTED])
On Sun, 7 Dec 2003, Kris Stark wrote:
Any ideas on how to do this one?
FWD requires an * on certain calls as a prefix character, but I cannot
seem to be able to get Prefix(*) to add that to the front of the
extension that is dialed... Setting up an extension that dials
(SIP/[EMAIL PROTECTED]) works just fine, but in trying to add the
prefix to a dialed number, it acts as if the Prefix command was not even
there, ie:
would dial [EMAIL PROTECTED], and not [EMAIL PROTECTED]
Any ideas?
Tnx
Kris
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Re: [Asterisk-Users] Prefix the * character
On Sun, 2003-12-07 at 23:57, Tilghman Lesher wrote:
On Sunday 07 December 2003 22:48, Kris Stark wrote:
Any ideas on how to do this one?
FWD requires an * on certain calls as a prefix character, but I
cannot seem to be able to get Prefix(*) to add that to the front of
the extension that is dialed... Setting up an extension that dials
(SIP/[EMAIL PROTECTED]) works just fine, but in trying to add the
prefix to a dialed number, it acts as if the Prefix command was not
even there, ie:
exten = 1,1,Prefix(*)
exten = *1,2,Dial(SIP/[EMAIL PROTECTED])
would dial [EMAIL PROTECTED], and not [EMAIL PROTECTED]
Any ideas?
Prefix is an older application which was more useful prior to being
able to manipulate variables (the days of BYEXTENSION instead of
${EXTEN}). Instead, do:
Dial(SIP/[EMAIL PROTECTED])
-Tilghman
Thanks! I actually had tried this to no avail, but then noticed after
digging through that it was in fact using the wrong entry to begin
with... An simpler match was being made, and so it never got to that
part of the dialplan...
Thanks again!
Kris
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[Asterisk-Users] Adtran 750
I have three quad FXS cards (in slots 1-3) and one quad FXO card in slot 4. I have a few telephone sets connected to the FXS cards, and * allows calling from one phone to another and from phone to console, etc. Diax also works. I have a CO line connected to one circuit of the the FXO card. When I dial from the console, the busy light on the FXO card comes on, but it doesn't break dial tone. When I put an Adtran FXS circuit in tone test mode, I can detect the tone at the punch-down block. When I put the FXO circuit in tone test mode and dial a local call, I hear the tone from *. However, I can't hear the tone at the punch-down block (and it hasn't broken dial tone.) The Adtran has been set to factory defaults (loop start, etc.) Is there something I'm missing here? Must the FXO cards occupy lower slots than FXS cards? Would the CO line originating at a pair-gain system (SLIC96?) require a different interface? TIA Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
