Re: [Asterisk-Users] voip-info.org DNS seems broken
Bill Reid wrote: For the last few days I can not resolve voip-info.org from many DNS servers. It does resolve with some DNS servers but I suspect it may be related more to caching. I've alerted James of the problems. I haven't seen them myself, so its hard for me to track. The wiki has become a too valuable resource for this community to continue to have these kind of problems. I assure you that James have spent a lot of time trying to solve the DNS and performance problems. Any help tracking the problems down is appreciated! We will check the caching timings in the zones. There's also an inconsistency between the zone and the DNS delegations from the .org TLD that will be fixed. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceMail Password problems
It seems to be a Grandstream specific thing. Prolly specific to certian GS firmware revs On Sat, 2003-12-13 at 15:18, Brian West wrote: I don't get why people always say dtmfmode=info mine works fine with rfc2833. bkw On Sat, 13 Dec 2003, Philipp von Klitzing wrote: Incorrect Password '4433211' for user '2000' (context =any) This is a FAQ: use dtmfmode=info in your sip.conf for your Grandstream Note: Don't forget to reload after modifying sip.conf. Cheers, Philipp ___ 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 -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to call from SNOM 200 to IP 7905G
Hello I have configured IP 7905G and SNOM 200 for Asterisk. Now problem is that I can call from IP 7905G to SNOM 200 but not the other way round. Instead I get FORBIDDEN Message on SNOM 200 LCD when ever I try to call IP7905 phone and asterisk generate following messages.. Please note 810, (129.82.44.222) are the phone number and IP of for my 7905G phone and 910, (129.82.44.226) are the phone number and IP for SNOM 200 in my configuration. Asterisk server IP is 129.82.44.215 Could you please help me in finding out the problem. /**/ Messages generated by Asterisk Server /**/ -- Executing Dial(SIP/910-f4ab, sip/810|30) in new stack -- Called 810 -- Got SIP response 404 Not Found back from 129.82.44.222 -- SIP/810-fafc is circuit-busy == Everyone is busy at this time WARNING[1209214400]: File pbx.c, Line 1160 (pbx_extension_helper): No application ' voicemail2' for extension (default, 810, 102) == Spawn extension (default, 810, 102) exited non-zero on 'SIP/910-f4ab' NOTICE[1125329600]: File chan_sip.c, Line 5298 (handle_request): Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '129.82.44.222' NOTICE[1125329600]: File chan_sip.c, Line 5298 (handle_request): Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '129.82.44.222' NOTICE[1125329600]: File chan_sip.c, Line 5298 (handle_request): Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '129.82.44.222' NOTICE[1125329600]: File chan_sip.c, Line 5298 (handle_request): Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '129.82.44.222' NOTICE[1125329600]: File chan_sip.c, Line 5298 (handle_request): Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '129.82.44.222' NOTICE[1125329600]: File chan_sip.c, Line 5298 (handle_request): Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '129.82.44.222' Regards Tony
Re: [Asterisk-Users] VoiceMail Password problems
Hi! I don't get why people always say dtmfmode=info mine works fine with rfc2833. bkw Dunno. I tried rfc2833 first, and had exactly the same problem as described below with voicemail (but only there). Info then worked just fine (as obviously also confirmed by this user here). Is there any other setup/setting that has influence on DTMF detection? Like NAT (yes for me) or anything else? However, more likely it's simply a GS firmware thing (4.17 on mine) - or production (hardware) issue with GS. Incorrect Password '4433211' for user '2000' (context =any) This is a FAQ: use dtmfmode=info in your sip.conf for your Grandstream Note: Don't forget to reload after modifying sip.conf. I am running GS Firmware 1.0.3.78 with Send DTMF = Via SIP INFO sip.conf for that phone is: [2001] type=friend username=2001 secret=test2 host=dynamic context=local-extensions Am able to access VM with no problem and use the phone via *-IAXtel to access other VM systems at USA toll-free numbers. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Party in Paris
Just to repeat - [EMAIL PROTECTED] Please reply if you wish. - Original Message - From: Stephen Wingfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, December 13, 2003 12:26 PM Subject: Re: [Asterisk-Users] * Party in Paris SATURDAY 20th I have had far fewer emails than the noise created earlier about Mark's arrival in Paris. Everyone who has contacted me I have replied to once. Again please - if you want to come please email me at [EMAIL PROTECTED] I will set a venue and time tomorrow evening to email to all concerned. Steve - Original Message - From: marrandy [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, December 11, 2003 7:58 PM Subject: Re: [Asterisk-Users] * Party in Paris On Thursday 11 December 2003 12:53 pm, Bob Knight wrote: Is the party at the Paris Hilton? sorry, couldn't help it... -- Bob Knight Bob...I'm really surprised at you !!! I thought you would have said, 'Is the party in Paris Hilton' lol ;-) -- Q: What's hard going in and soft and sticky coming out? A: Chewing gum. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voip-info.org DNS seems broken
If you want a mirror of you site, I can give some space (how much is needed) , plus mysql,php, blah blah blah For free, of course. Matteo. Il dom, 2003-12-14 alle 10:00, Olle E. Johansson ha scritto: Bill Reid wrote: For the last few days I can not resolve voip-info.org from many DNS servers. It does resolve with some DNS servers but I suspect it may be related more to caching. I've alerted James of the problems. I haven't seen them myself, so its hard for me to track. The wiki has become a too valuable resource for this community to continue to have these kind of problems. I assure you that James have spent a lot of time trying to solve the DNS and performance problems. Any help tracking the problems down is appreciated! We will check the caching timings in the zones. There's also an inconsistency between the zone and the DNS delegations from the .org TLD that will be fixed. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voip-info.org DNS seems broken
For the last few days I can not resolve voip-info.org from many DNS servers. It does resolve with some DNS servers but I suspect it may be related more to caching. I've alerted James of the problems. I haven't seen them myself, so its hard for me to track. The wiki has become a too valuable resource for this community to continue to have these kind of problems. I assure you that James have spent a lot of time trying to solve the DNS and performance problems. Any help tracking the problems down is appreciated! If it helps, I'm willing for our company to offer secondary DNS. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voip-info.org DNS seems broken
Thank you for all the offers. I think the list would prefer if you mailed me off-list :-) A gentle remark, no attack. You are really most kind to offer your assistance. And if any other users have any problems and can track them down, I would appreciate you mailing me so we can make the Wiki available to everyone all the time. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voip-info.org DNS seems broken
All, Please excuse my last post. I meant to send to Olle directly (off-list), and was operating on not enough sleep. Of course, I picked the best message of all to send back to the list. Thanks, Pat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Patrick Cantwell Sent: Sunday, December 14, 2003 9:26 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] voip-info.org DNS seems broken Ollie, It's been my experience that you can never have too many mirrors and/or at least secondary DNS. If you'd like me to do either for you, I'd be more than glad. Thanks! Pat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Olle E. Johansson Sent: Sunday, December 14, 2003 6:37 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] voip-info.org DNS seems broken Thank you for all the offers. I think the list would prefer if you mailed me off-list :-) A gentle remark, no attack. You are really most kind to offer your assistance. And if any other users have any problems and can track them down, I would appreciate you mailing me so we can make the Wiki available to everyone all the time. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax parsing error
Hi, I'm getting this error and don't know how to fix it. IAX2 seems to work though... [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found WARNING[1074439936]: File chan_iax2.c, Line 5465 (set_config): Ignoring port for now == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 4 Thanks Hector. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and fwd
Hi, could anyone please provide a working sample of how to configure asterisk to connect to fwd? I've tried the one at www.loligo.com and it doesn't work. Not even when calling to 5. Can you advise on how to debug sip (or trace and view sip packets) from the asterisk server to fwd so we can try to understand what exactly is not working? Please be advised that our asterisk server is behind a NAT firewall. Thanks in advance, Tom.
Re: [Asterisk-Users] iax parsing error
I'm getting this error and don't know how to fix it. IAX2 seems to work though... [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found WARNING[1074439936]: File chan_iax2.c, Line 5465 (set_config): Ignoring port for now == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 4 I think this might be related to port=5036 in iax.conf, knowing that iax2 uses port 4569. If its there, comment it out and reload. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and fwd
If U want I can send U my settings. My FWD is working fine. Let me know... Chris HARIGA - Original Message - From: Shoval Tomer To: [EMAIL PROTECTED] Sent: Sunday, December 14, 2003 10:31 AM Subject: [Asterisk-Users] Asterisk and fwd Hi, could anyone please provide a working sample of how to configure asterisk to connect to fwd? I've tried the one at www.loligo.com and it doesn't work. Not even when calling to 5. Can you advise on how to debug sip (or trace and view sip packets) from the asterisk server to fwd so we can try to understand what exactly is not working? Please be advised that our asterisk server is behind a NAT firewall. Thanks in advance, Tom.
[Asterisk-Users] CAMA MF signaling for a 911 Trunk
Hi all, I'm trying to get a handoff between me and a carrier going using Asterisk. I need to handoff using CAMA signaling. On a Cisco, you can see the configuration types that I'm referring to on this site as an example: http://www.cisco.com/en/US/products/hw/routers/ps221/prod_configuration_guide09186a008019b16e.html#35393 The example of the interaction is the following: CLEC end office: Seizure 9-1-1 Tandem: Wink CLEC end office: KP (Keypulse) NPA ST (Start) 9-1-1 Tandem: Wink CLEC end office: KP I (Info Digit) NXX ST The call is originating from the VoIP side. I have calling and called number in the SIP invite. For the life of me, I can't figure out how to pulse out the digits in this format to make the signaling compliant with the interface. Any suggestions would be appreciated... William _ Tired of slow downloads and busy signals? Get a high-speed Internet connection! Comparison-shop your local high-speed providers here. https://broadband.msn.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Two Stage Dialing for MF CAMA trunk
Hi all, I am trying to setup a ZAP interface to do MF signaling for a handoff to a 911 tandem. The signaling I need to perform on the T1 is this: 9-1-1 Tandem: Wink CLEC end office: KP (Keypulse) NPA ST (Start) 9-1-1 Tandem: Wink CLEC end office: KP I (Info Digit) NXX ST As I'm not as familiar with the Zaptel configurabliity, I'm not really sure how to do this. Do I dial twice or something similar in the dial plan? Thanks, William _ Shop online for kidsÂ’ toys by age group, price range, and toy category at MSN Shopping. No waiting for a clerk to help you! http://shopping.msn.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)
- Original Message - From: "Alastair Maw" [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, December 12, 2003 4:58 PM Subject: Re: [Asterisk-Users] FAX, IAX and *Maybe I'm dreaming...:-) On 12/12/03 13:56, Dan wrote: This is because the fax is transmitted using the audio stream. It is not related to the signaling protocol (SIP/IAX etc.) but to the audio codec used. Fax uses FSK modulation to transmit the data. If you compress this in a lossy way (GSM, MP3, whatever) then the integrity of the data is affected (more or less seriously depending on the codec used). Fax machines are generally quite picky, so compressing faxes is unlikely to work. I'm wondering why on earth you want to push fax data over a VoIP link at all. Fax compression isn't very efficient. Who wants that??? By fax data I mean the data contained in a fax (basically a picture file), not the fax data audio stream. It can be converted (GIF or JPG) then sent reliable over a slow IP link. Just a special codec at both ends, able to pass the data to the fax app or a fax machine connected to a TDM400/ AT or whatever. It would be much less bandwidth intensive to decode the fax and send it over as proper data rather than audio, compressed using gzip/gif/png/something else. This is exactly what would be great to have it. The question asked here, "why on earth you want to push fax data over a VoIP link atall. Fax compression isn't very efficient." may speak volumes about the future role of VOIP. My plans are to role out a VOIP connection to thousands of Customers. Many have legacy fax equipment. Am I to assume that they will toss out their fax equipment and join the PC based faxing crowd? I don't think I can control this. If I am going to offer an aternative to the legacy wire providers then I have to offer a comparable service. One that for example allows a customer to use a legacy fax machine in the same way. If this thought sidetracks the intent of this thread, you have my apologies, but I do think that legacy fax functionality is essential. Jeff
RE: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ProvoCityPower Sent: Sunday, December 14, 2003 12:51 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FAX, IAX and *Maybe I'm dreaming...:-) The question asked here, why on earth you want to push fax data over a VoIP link at all. Fax compression isn't very efficient. may speak volumes about the future role of VOIP. My plans are to role out a VOIP connection to thousands of Customers. Many have legacy fax equipment. Am I to assume that they will toss out their fax equipment and join the PC based faxing crowd? I don't think I can control this. If I am going to offer an aternative to the legacy wire providers then I have to offer a comparable service. One that for example allows a customer to use a legacy fax machine in the same way. If this thought sidetracks the intent of this thread, you have my apologies, but I do think that legacy fax functionality is essential. Jeff I think what you're talking about here is an absolute necessity if VOIP is ever to compete for traditional analog service. A lot of users will not want to change their ways. As stupid as I think FAXing and FAX machines are there are still millions of people who prefer to send and recive a skewed low resolution fax on thermal paper than the transmit and receive them over a computer. Even though, I think FAXing is as ancient as sending a letter through the mail it needs to be supported however the end user wishes to use it. With that being said, I think it simply comes down to the codecs being used. I seem to recall that alaw and ulaw were acceptable codecs, though I know nothing about this first hand. The other option is some sort of FAX proxy, though that seems a little too complicated if you ask me. -Bill ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and fwd
Shoval Tomer wrote: Hi, could anyone please provide a working sample of how to configure asterisk to connect to fwd? I've tried the one at www.loligo.com http://www.loligo.com/ and it doesn't work. Not even when calling to 5. Check the Asterisk FAQ at http://www.voip-info.org Can you advise on how to debug sip (or trace and view sip packets) from the asterisk server to fwd so we can try to understand what exactly is not working? Test the CLI command sip debug. Regards, /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voice mail - sip:notify message
At 12:29 AM -0800 12/13/03, SW wrote: Hi folks, To provide MWI, * will send out a sip:notify message to the UA. The originator of this message is asterisk, as shown below; NOTIFY sip:[EMAIL PROTECTED]:5065 SIP/2.0 Via: SIP/2.0/UDP 66.121.xxx.yyy:5060;branch=z9hG4bK0466cb21 From: asterisk sip:[EMAIL PROTECTED];tag=as0ffb1bdc === To: sip:[EMAIL PROTECTED]:5065 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 38 Messages-Waiting: yes Voicemail: 4/11 Is there a way that I can change this 'originator' to a numeric value ? Cheers SW SW - See the below conversation from April regarding the same topic. I don't think this became a patch or made it into CVS. JT At 7:59 PM -0500 4/27/03, Mark Spencer wrote: Sounds like the SNOM expects to use our Contact to get a hold of us. It should be simple to add something like voicemail=foo in the general section for setting the voicemail extension to use in the contact area. Mark On Thu, 24 Apr 2003, WipeOut . wrote: Here is the trace if anyone is interested.. NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK523b1b63 From: asterisk sip:[EMAIL PROTECTED];tag=as3da6a846 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 36 Message-Waiting: yes Voicemail: 1/0 Hi, The MWI is working on the SNOM 200 but the problem is that when you press the MWI button it attempts to dial asterisk sip:[EMAIL PROTECTED] where 192.168.1.200 is the IP address of my * box. How can I modify this so the return path is correct, which on my setup is extension 8500 for voicmailmain?? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WiSip phone experiences?
http://www.loligo.com/asterisk/misc/WiSIP/ Works decently enough. Still some software bugs to work out. JT At 10:10 AM -0600 12/12/03, Michael Graves wrote: Anyone here have any good/bad things to say about first hand experience with the new Wifi SIP phones? I am considering one for my office as an alternative to FXS+Analog cordless. Thanks, Michael Graves -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] FWD 54245 I've been everywhere man, I've been everywhere. - Hank Snow ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WiSip phone experiences?
thats an old review jeff pulver said firmware was better now, is it worth 250USD? im thinking of one does it autodetect ssid? works with low signal? sound quality? Miguel On Sun, 2003-12-14 at 18:34, John Todd wrote: http://www.loligo.com/asterisk/misc/WiSIP/ Works decently enough. Still some software bugs to work out. JT At 10:10 AM -0600 12/12/03, Michael Graves wrote: Anyone here have any good/bad things to say about first hand experience with the new Wifi SIP phones? I am considering one for my office as an alternative to FXS+Analog cordless. Thanks, Michael Graves -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] FWD 54245 I've been everywhere man, I've been everywhere. - Hank Snow ___ 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] Two Stage Dialing for MF CAMA trunk
Looking at the code in chan_zap.c, I only see options for feature group B and feature group D MF. The 2 stage MF signalling you are asking for isn't implmented in the latest asterisk source code. I would suggest you post a feature request detailing your 2 stage dialing requirement to bugs.digium.com as 911 support is something which could be useful to others. Steve. On Sunday 14 December 2003 09:53, William Flanagan wrote: Hi all, I am trying to setup a ZAP interface to do MF signaling for a handoff to a 911 tandem. The signaling I need to perform on the T1 is this: 9-1-1 Tandem: Wink CLEC end office: KP (Keypulse) NPA ST (Start) 9-1-1 Tandem: Wink CLEC end office: KP I (Info Digit) NXX ST As I'm not as familiar with the Zaptel configurabliity, I'm not really sure how to do this. Do I dial twice or something similar in the dial plan? Thanks, William _ Shop online for kids toys by age group, price range, and toy category at MSN Shopping. No waiting for a clerk to help you! http://shopping.msn.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] modem data calls through FXS / FXO digium cards failing
Hi guys, I think I posted on this issue before, but didn't get a response. I've still not been able to resolve the issue. I've got a small installation of Asterisk running one 4 port FXS Digium card and 1 FXO Digium card. I'm having difficulty routing modem call through one of the extensions out through the FXO card. By difficulty, I mean it won't work. The calls won't even sync up. If I disconnect the phone line from the FXO card and run the modem directly through that (completely bypassing Asterisk), the calls go through fine as usual. Is there some sort of line quality I can specify for certain extensions to allow data calls to better pass through? Any options I can specify on the Dial application that boost the quality or drop echo cancelling or are these strategies even going to do me any good? Does anyone have a similar hardware configuration working? Thanks. jl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX 1/2 registration
Hi, How can I do to register an Asterisk server using just IAX2? If I have a line like the following in iax.conf register = user:[EMAIL PROTECTED] the server tries to register with both IAX and IAX2. if the line is : register = user:[EMAIL PROTECTED]:4569 then the same, tries both on the same port (means IAX(1) on 4569 too). This can be shown using iax show registry It is possible to disable registration requests on IAX(1)? As IAXTEL does not support IAX1 anymore, I still have registration requests in IAX1.. Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX 1/2 registration
Just noload chan_iax right? Mark On Sun, 14 Dec 2003, Dan wrote: Hi, How can I do to register an Asterisk server using just IAX2? If I have a line like the following in iax.conf register = user:[EMAIL PROTECTED] the server tries to register with both IAX and IAX2. if the line is : register = user:[EMAIL PROTECTED]:4569 then the same, tries both on the same port (means IAX(1) on 4569 too). This can be shown using iax show registry It is possible to disable registration requests on IAX(1)? As IAXTEL does not support IAX1 anymore, I still have registration requests in IAX1.. Thanks, Dan ___ 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] IAX 1/2 registration
Hi, From: Mark Spencer [EMAIL PROTECTED] Just noload chan_iax right? Mark Yes, but... I still need IAX(1) till the problem with DIAX will be solved. When a call is placed through Asterisk using IAX2 and DIAX, the call is automatically droped apparently without any reason, after about 60'. It works ok with IAX(1). There is no other way to disable IAX(1) registration? Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] outbound dialing / wait for keypress?
hi there. i've got a question about outbound dialing. here's my scenario: 1. i build a list of phone numbers from a database 2. when a call comes in, i begin dialing from the list 3. when an outbound call is answered, i connect the caller to that line. so far, i'm able to do this with an agi script to dynamically build a dialplan. i make repeated use of this perl call: system(asterisk -r -x 'add extension 100,++$i,Dial(Zap/2/[number],15) into local; to create a dialplan that looks like this: s,1,Answer s,2,agi,script.agi s,3,Goto(local,100,1) 100,1,Dial(Zap/2/[number],15) 100,2,Dial(Zap/2/[another_number],15) ... so far, so good - if the first number dialed is busy or times out, it tries dialing the next number, and so on until one is answered at which point the incoming and outgoing calls are connected. my problem is how to handle the case where an outbound call is answered by, say, an answering machine. ideally, i'd like Dial to work such that it can be configured to play a recording and wait for a keypress before considering the call answered. however, this doesn't seem to be the case. i imagine that someone else has already encountered this problem - if so, does anyone know what the solution is (i've already checked the various resources, and haven't found anything). if not, does anyone have thoughts on how best to approach this? thanks, tad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ignorepat
Hi I have the following configuration at home one ZAPTEL interface connecting to an FXO card and two SIP UAs connecting to asterisk locally. I have configured extensions.conf such that dialing 9 on the SIP phones allows me to dial an outbound number via the FXO interface . Works fine. What's not working is that pressing 9 should causes either GS BT-100 phone to reacquire a dialtone since I have placed ignorepat = 9 in the config file. Any ideas? rgds burak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX 1/2 registration
Hi, Citeren Mark Spencer [EMAIL PROTECTED]: Just noload chan_iax right? Mark Hmm, I understand there is little or no desire to really work on this type of issue, but noload chan_iax is slightly too rigorous for me. Consider this scenario: I prefer using IAX2 with every partner I can talk to (IAXtel, uplinks) but I would still like users to be able to use IAX1 as a backward-compatible method. Therefore, noload chan_iax is not desirable. Suggestion for those with this issue: Make the firewall (iptables) on your asterisk box drop or reject the IAX1 packets toward destinations where we do not want IAX1 to be used :-) -- Best regards, Florian Overkamp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voice mail - sip:notify message
Thanks, John. My requirement here is little different. I am using * as a Voice Mail server for Vocal ((in addition It does codec conversion and routing to PSTN/SIP and PSTN/H323). Thing is Vocal doesn't seems to like the Notify message coming from user asterisk. If I can modify this I will have * seamlessly providing full capabilities of voice mail to Vocal . I can't find a way to get the VMWI to pass through to a SIP client of Vocal. Cheers SW = Message: 6 Date: Sun, 14 Dec 2003 13:33:09 -0500 To: [EMAIL PROTECTED] From: John Todd [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] voice mail - sip:notify message Reply-To: [EMAIL PROTECTED] At 12:29 AM -0800 12/13/03, SW wrote: Hi folks, To provide MWI, * will send out a sip:notify message to the UA. The originator of this message is asterisk, as shown below; NOTIFY sip:[EMAIL PROTECTED]:5065 SIP/2.0 Via: SIP/2.0/UDP 66.121.xxx.yyy:5060;branch=z9hG4bK0466cb21 From: asterisk sip:[EMAIL PROTECTED];tag=as0ffb1bdc === To: sip:[EMAIL PROTECTED]:5065 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 38 Messages-Waiting: yes Voicemail: 4/11 Is there a way that I can change this 'originator' to a numeric value ? Cheers SW SW - See the below conversation from April regarding the same topic. I don't think this became a patch or made it into CVS. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Crash System Command
When I run the asterisk System command my asterisk crashes. When I monitor the console this is the error I get. Any suggestions? exten = 1,1,System(ls) this is the error I get: [EMAIL PROTECTED] asterisk]# Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and fwd
Please do. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris HARIGA Sent: Sunday, December 14, 2003 8:41 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk and fwd If U want I can send U my settings. My FWD is working fine. Let me know... Chris HARIGA - Original Message - From: Shoval Tomer To: [EMAIL PROTECTED] Sent: Sunday, December 14, 2003 10:31 AM Subject: [Asterisk-Users] Asterisk and fwd Hi, could anyone please provide a working sample of how to configure asterisk to connect to fwd? I've tried the one at www.loligo.com and it doesn't work. Not even when calling to 5. Can you advise on how to debug sip (or trace and view sip packets) from the asterisk server to fwd so we can try to understand what exactly is not working? Please be advised that our asterisk server is behind a NAT firewall. Thanks in advance, Tom.
[Asterisk-Users] GS early dial
Am I assuming that a GS set to early dial to * dosn't work. Or am I missing something? Tried inband, info and rfc288, all nojoy. I'm assuming that it's not/supported or GS bug, only asking because it's assumptions that alwas get me :-) GS firmware 1.0.4.26 Thanx in advance John Breeden Hawaii
Re: [Asterisk-Users] GS early dial
it doesnt work here, same firmware 4.26 tryed it with 4.18 also and it doesnt work, i press any number and it gets screw up i will try it with the handytone ata286 and see if it works, anyway its the same firmware but its worth to try out Miguel On Sun, 2003-12-14 at 21:13, John Breeden wrote: Am I assuming that a GS set to early dial to * dosn't work. Or am I missing something? Tried inband, info and rfc288, all nojoy. I'm assuming that it's not/supported or GS bug, only asking because it's assumptions that alwas get me :-) GS firmware 1.0.4.26 Thanx in advance John Breeden Hawaii ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)
On Sun, 2003-12-14 at 12:18, [EMAIL PROTECTED] wrote: From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ProvoCityPower The question asked here, why on earth you want to push fax data over a VoIP link at all. Fax compression isn't very efficient. may speak volumes about the future role of VOIP. My plans are to role out a VOIP connection to thousands of Customers. Many have legacy fax equipment. Am I to assume that they will toss out their fax equipment and join the PC based faxing crowd? I don't think I can control this. If I am going to offer an aternative to the legacy wire providers then I have to offer a comparable service. One that for example allows a customer to use a legacy fax machine in the same way. I think what you're talking about here is an absolute necessity if VOIP is ever to compete for traditional analog service. A lot of users will not want to change their ways. As stupid as I think FAXing and FAX machines are there are still millions of people who prefer to send and recive a skewed low resolution fax on thermal paper than the transmit and receive them over a computer. Even though, I think FAXing is as ancient as sending a letter through the mail it needs to be supported however the end user wishes to use it. With that being said, I think it simply comes down to the codecs being used. I seem to recall that alaw and ulaw were acceptable codecs, though I know nothing about this first hand. The other option is some sort of FAX proxy, though that seems a little too complicated if you ask me. Did DVD players have to accommodate VHS tapes? Did VHS players have to accept beta? Why does VoIP have to deal with an accent protocol that can't handle lossy audio, nor irregular delays? Also why should we be soo wasteful when fax machines need a 80K codec to get the data across IP, and the faster machines I see say 15 secs per page. So why should we send 1.2meg when 150k is fine? Also who says Fax should ever be required on IP? My office has been using VoIP for all voice traffic for over a year now, but always left the fax machine on a analog line. The analog line was cheap enough to not be a concern. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] unable to configure my Grandstream phone
Hi All, i received my X100P and Grandstream phone last week. i started configuring my * and with the help of ur mailing lists i was able to configure it. (when ever i got struck i searched this list and found my answer. thanks a lot and this list is awesome). i still hv a small problem and hope someone could help me out. This is my setup. RH 7.2 serving as my * server. i hv got couple of my laptops and desktops running MSN 4.7. I hv installed and configured X100P and Grandstream phone the following configurations are working MSN Msgr - * - MSN Msgr MSN Msgr - * - X100P - PSTN PSTN - * - MSN Msgr PSTN - * - Grandstream (pl note) the following are *not* working MSN Msgr - * - Grandstream Grandstream - * - MSN Msgr GrandStream - * - PSTN PSTN - * - Grandstream the error i am getting in this case is - Attempting native bridge of SIP/2003-b895 and SIP/2000-53e2WARNING[5126]: File chan_sip.c, Line 1954 (process_sdp): No compatible codecs! -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 192.168.0.58 and then the call drops. When i am making a call using Grandstream ph, it rings the other side when they pick up the phone the call then drops. then i get the above error message. the follwoing us sip and Grandstream conf [general]port = 5060bindaddr = 0.0.0.0context = bogon-calls;context = defaultdisallow=allallow=gsm [2000]; Grandstream phone type=friendusername=2000secret=qweqwehost=dynamiccontext=from-sipmailbox=2000dtmfmode=inband [2002] type=friendhost=dynamicinsecure=yesdtmfmode=inband;dtmfmode=rfc2833context=from-sipmailbox=2002;auth=plaintext [2003] type=friendhost=dynamicinsecure=yesdtmfmode=inbandcontext=from-sipmailbox=2003 ;[2000] ;type=friend;username=2000;secret=qweqwe;auth=md5;host=dynamic;context=from-sip;dtmfmode=inband;mailbox=2000 [2001] type=friendusername=2001secret=asdasdauth=md5host=dynamiccontext=from-sipdtmfmode=inbandmailbox=2001 Grandstream configuration details SIP Server: 192.168.0.4 (my * box)SIP Userid: 2000 (userid is same as extensionAuthenticate ID: 2000Authenticate password: qweqwe Send DTMF: Via SIP info (in order for the dtmf to be recognized byvoicemail) Program--1.0.4.17 Bootloader--1.0.0.11 HTML--1.0.0.19 any idea why my Grandstream drops the calls. thanks a lot and appreciate ur help. -B Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing
Re: [Asterisk-Users] unable to configure my Grandstream phone
- Original Message - From: Balaji NJL To: [EMAIL PROTECTED] Sent: Monday, December 15, 2003 8:47 AM Subject: [Asterisk-Users] unable to configure my Grandstream phone snip Attempting native bridge of SIP/2003-b895 and SIP/2000-53e2 WARNING[5126]: File chan_sip.c, Line 1954 (process_sdp): No compatible codecs! -- Got SIP response 481 Call Leg/Transaction Does Not Exist back from 192.168.0.58 and then the call drops. When i am making a call using Grandstream ph, it rings the other side when they pick up the phone the call then drops. then i get the above error message. the follwoing us sip and Grandstream conf [general] port = 5060 bindaddr = 0.0.0.0 context = bogon-calls ;context = default disallow=all allow=gsm Balaji, Grandstreams do not support GSM. Options available can be seen on the GS config page. Unless you purchase G729s (for low bandwidth), your only choice is G711. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] nexthop: free service to pstn via fwd/iaxtel
(found on dslr voip forum) http://www.freephoneproject.com/nexthop/ I think this is pretty cool. not too sure how stable/durable though, at least not until there are a lot more nodes providing service. also unsure of the legality vs. your average telco's consumer tos, but it's definitely not reselling. jeff pulver (crom bless him) seemed to be fairly amused, stuff like this fits right into his vision as far as I can tell. looking forward to the day when enough people have hardphones that non-free pstn access becomes fairly redundant. cheers ++dg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk problem
Dear All,
I am a new user of Asterisk
interested in setting up a VOIP network based on Asterisk. I have deployed a few
Asterisk servers running on T400P and have started a few weeks ago to run some
LIVE traffic on one of the servers. Most of my current traffic is via H323 to
and from other carriers / customers and I am using the H323 driver written by
Jeremy of Nufone.
I have had problems with
this, as it seems that as time goes, for instance after 5 or 6 hours or
sometimes 3 to 4 hours or less, the system would not be able to take any
incoming calls via H323 protocol. All the calls from my H323 customers (they are
mostly carriers sending traffic to me from other gateways such as cisco AS5300)
would be rejected and the system just would not take the calls for some reasons.
I could, however, call into the PRI (Zap interface) and make an outbound H323
calls to other gateways of my carriers. That is, incoming H323 calls would not
work after a certain period of operation time but outgoing H323 calls would
still be possible.
I have then downloaded newer
version of the Asterisk about 10 days ago, but then my Asterisk would start to
crash on me, the module would just stop running by itself and I had to restart
the Asterisk. Sometimes, it would just stop running, but 75% of the time, I
would see this error message:
Connected to Asterisk CVS-12/06/03-03:06:28
currently running on localhost (pid = 23720) -- Remote
UNIX connectionlocalhost*CLI /usr/sbin/safe_asterisk: line 6: 23720
Segmentation fault asterisk ${ASTARGS}
1/dev/${TTY} /dev/${TTY}Asterisk ended with exit status
139Asterisk exited on signal 11.Automatically restarting
Asterisk.
Disconnected from Asterisk
serverExecuting last minute cleanups[EMAIL PROTECTED] asterisk]#
/usr/sbin/safe_asterisk: line 6: 23752 Segmentation
fault asterisk ${ASTARGS} 1/dev/${TTY}
/dev/${TTY}Asterisk ended with exit status 139Asterisk exited on
signal 11.Automatically restarting Asterisk.
It seems that it would
start to restart by would fail, and then I had to restart it manually. This
Segmentation error happens only with new version of Asterisk but not with older
version (10 days ago), does anyone here have any of such experience or know why
this is happening? Is this because I am running T400P instead of TE410P hardware
and that the newer version of software is not 100% compatible with the old
hardware? The inbound H323 problem, however, exists for both the newer and older
version of Asterisk, and I wonder if anyone has had this experience before. Is
this pertaining to the hardware as well or is there any other
reason/
Ladies and Gentlemen, any
feedbacks and advices would be appreciated. Many
Thanks
Tommy Chan
[Asterisk-Users] Cisco Gateway Integration
Title: Message Has anyone succesfully integrated * with a cisco voice gateway ?
[Asterisk-Users] Cisco 7960 lockups - any experiences?
This is almost certainly not an Asterisk-specific posting, but due to my inability to find a VoIP-focused Cisco list, I'll post here in the hopes of finding a more diverse user community. I am using a Cisco 7960 (version 6.0 SIP firmware) with Asterisk, and have been experiencing situations where the phone locks up. Locks up means that the bottom part of the screen (Your current options and the redial/newcall/cfwdall keys) disappears, and all keys on the keypad are non-functional except for the *-6-settings reboot keys. Previous software (4.4) exhibited the same symptoms. I replaced the 7960 with a brand new 7960G, thinking it was bad hardware. Same symptoms. I replaced the power supply, thinking that perhaps could be the problem Same symptoms. I upgraded to 6.0. Same symptoms. I replaced the ethernet cables. Same symptoms. I invoked an ancient curse-removal spell involving chicken bones and eye of newt over the phone. Same symptoms. A phone elsewhere in the same office (same switch, etc) works with no problems, but does not have a PC hooked to the 'PC' ethernet port. In fact, the first 7960 I tried, with the swapped out power supply, works well elsewhere. I have used 79xx boxes in many circumstances, and all have performed admirably. However, I rarely have configured them so that a PC is connected to the other side of the device. The PC is running Windows XP, and sees mild SSH/email/web traffic. I cannot say if there is a correlation between traffic volume and failure intervals; I have no data. The only thing that remains the same is the PC on one side, and the switch port on the other. Previously, the PC attached to that switch port had worked without any problems for quite some time before I put the 7960 in place. Even if the problem is the PC or the switchport, a failure or malfunction on one of those two components should not cause the phone to malfunction. If it is a problem with the 7960 pushing traffic across the tiny built-in switch, then we have a real problem on our hands that Cisco had better fix, pronto. I would appreciate reports of others as to similar issues or resolutions which they found effective. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] unable to configure my Grandstream phone
Hi Paul, thanks for the quick response. i tried the following configuration / combination still no luck [general] port = 5060 bindaddr = 0.0.0.0 context = bogon-calls ;context = default disallow=all allow=g729 allow=gsm allow=ulaw when i tried g711 i am getting an error in * that codec not found. When i specify only g729 my MSN doesnt work. thanks, -B - Original Message - From: Paul Liew [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, December 14, 2003 2:14 PM Subject: Re: [Asterisk-Users] unable to configure my Grandstream phone - Original Message - From: Balaji NJL To: [EMAIL PROTECTED] Sent: Monday, December 15, 2003 8:47 AM Subject: [Asterisk-Users] unable to configure my Grandstream phone snip Attempting native bridge of SIP/2003-b895 and SIP/2000-53e2 WARNING[5126]: File chan_sip.c, Line 1954 (process_sdp): No compatible codecs! -- Got SIP response 481 Call Leg/Transaction Does Not Exist back from 192.168.0.58 and then the call drops. When i am making a call using Grandstream ph, it rings the other side when they pick up the phone the call then drops. then i get the above error message. the follwoing us sip and Grandstream conf [general] port = 5060 bindaddr = 0.0.0.0 context = bogon-calls ;context = default disallow=all allow=gsm Balaji, Grandstreams do not support GSM. Options available can be seen on the GS config page. Unless you purchase G729s (for low bandwidth), your only choice is G711. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ignorepat
Sip phones generate their own dialtone. The ignore pat option is meaningless with regard to SIP phones. I would check the Qrandstream's dialplan and see if you can program it to ignore the dialtone after a '9' is pressed. I had to do something similar for my Sipura SPA-2000. Steve. On Sunday 14 December 2003 12:18, Burak Balasaygun wrote: Hi I have the following configuration at home one ZAPTEL interface connecting to an FXO card and two SIP UAs connecting to asterisk locally. I have configured extensions.conf such that dialing 9 on the SIP phones allows me to dial an outbound number via the FXO interface . Works fine. What's not working is that pressing 9 should causes either GS BT-100 phone to reacquire a dialtone since I have placed ignorepat = 9 in the config file. Any ideas? rgds burak ___ 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] FAX, IAX and *....Maybe I'm dreaming...:-)
ProvoCityPower wrote: The question asked here, why on earth you want to push fax data over a VoIP link at all. Fax compression isn't very efficient. may speak volumes about the future role of VOIP. My plans are to role out a VOIP connection to thousands of Customers. Many have legacy fax equipment. Am I to assume that they will toss out their fax equipment and join the PC based faxing crowd? I don't think I can control this. If I am going to offer an aternative to the legacy wire providers then I have to offer a comparable service. One that for example allows a customer to use a legacy fax machine in the same way. If this thought sidetracks the intent of this thread, you have my apologies, but I do think that legacy fax functionality is essential. Legacy FAX will be very important for years. The last people to abandon it are the most senior managers. Therefore, apart from anything else, their buyin to a change to VoIP depends on keeping their olde worlde FAX facilities alive. However, I think this has nothing to do with the original poster's intent. Sending FAXes over IP as audio is dumb. Its troublesome, error prone, and consumes too much bandwidth. The right way is to use a FAX modem to translate between audio and the digital data stream of the image itself. Then the compact image data can be conveyed reliably. If the far end is not using IP, another modem can be used to return the FAX to its analogue form for delivery. If you wish to interwork traditional analogue FAX machines, with newer IP capable FAX machines you *have* to do this. This is what the H323 protocols do. T.38 is the relevant FAX over IP protocol. I don't know if there is a similar SIP standard, but there should be. FAX to e-mail and e-mail to FAX is the other inportant way to blend the old with the new. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] unable to configure my Grandstream phone
i hv also added the alaw [general] port = 5060 bindaddr = 0.0.0.0 context = bogon-calls ;context = default disallow=all allow=g729 allow=gsm allow=ulaw allow=alaw now i am able to call from my MSN - * -GS but the other way is not working. i am getting lot of noise when i try to place any call from my GS. i am no longer getting codec not compatible error message anymore. i am still unable to place any calls using my GS (to my internal MSN extensions or to external PSTN). thanks for ur help, -B - Original Message - From: Balaji NJL [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Sunday, December 14, 2003 3:40 PM Subject: Re: [Asterisk-Users] unable to configure my Grandstream phone Hi Paul, thanks for the quick response. i tried the following configuration / combination still no luck [general] port = 5060 bindaddr = 0.0.0.0 context = bogon-calls ;context = default disallow=all allow=g729 allow=gsm allow=ulaw when i tried g711 i am getting an error in * that codec not found. When i specify only g729 my MSN doesnt work. thanks, -B - Original Message - From: Paul Liew [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, December 14, 2003 2:14 PM Subject: Re: [Asterisk-Users] unable to configure my Grandstream phone - Original Message - From: Balaji NJL To: [EMAIL PROTECTED] Sent: Monday, December 15, 2003 8:47 AM Subject: [Asterisk-Users] unable to configure my Grandstream phone snip Attempting native bridge of SIP/2003-b895 and SIP/2000-53e2 WARNING[5126]: File chan_sip.c, Line 1954 (process_sdp): No compatible codecs! -- Got SIP response 481 Call Leg/Transaction Does Not Exist back from 192.168.0.58 and then the call drops. When i am making a call using Grandstream ph, it rings the other side when they pick up the phone the call then drops. then i get the above error message. the follwoing us sip and Grandstream conf [general] port = 5060 bindaddr = 0.0.0.0 context = bogon-calls ;context = default disallow=all allow=gsm Balaji, Grandstreams do not support GSM. Options available can be seen on the GS config page. Unless you purchase G729s (for low bandwidth), your only choice is G711. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 lockups - any experiences?
So do I have this right? You have a 7960 hooked up to your network. You have a PC plugged into the second Ethernet port of the phone. This 7960 freezes. If you move the phone somewhere else in the office, without the PC, it doesn't freeze? Does it freeze when it's in the same location, but the PC isn't plugged into it? That is, does this happen only when a PC is plugged into the phone? Paul Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Sunday, December 14, 2003 2:59 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960 lockups - any experiences? This is almost certainly not an Asterisk-specific posting, but due to my inability to find a VoIP-focused Cisco list, I'll post here in the hopes of finding a more diverse user community. I am using a Cisco 7960 (version 6.0 SIP firmware) with Asterisk, and have been experiencing situations where the phone locks up. Locks up means that the bottom part of the screen (Your current options and the redial/newcall/cfwdall keys) disappears, and all keys on the keypad are non-functional except for the *-6-settings reboot keys. Previous software (4.4) exhibited the same symptoms. I replaced the 7960 with a brand new 7960G, thinking it was bad hardware. Same symptoms. I replaced the power supply, thinking that perhaps could be the problem Same symptoms. I upgraded to 6.0. Same symptoms. I replaced the ethernet cables. Same symptoms. I invoked an ancient curse-removal spell involving chicken bones and eye of newt over the phone. Same symptoms. A phone elsewhere in the same office (same switch, etc) works with no problems, but does not have a PC hooked to the 'PC' ethernet port. In fact, the first 7960 I tried, with the swapped out power supply, works well elsewhere. I have used 79xx boxes in many circumstances, and all have performed admirably. However, I rarely have configured them so that a PC is connected to the other side of the device. The PC is running Windows XP, and sees mild SSH/email/web traffic. I cannot say if there is a correlation between traffic volume and failure intervals; I have no data. The only thing that remains the same is the PC on one side, and the switch port on the other. Previously, the PC attached to that switch port had worked without any problems for quite some time before I put the 7960 in place. Even if the problem is the PC or the switchport, a failure or malfunction on one of those two components should not cause the phone to malfunction. If it is a problem with the 7960 pushing traffic across the tiny built-in switch, then we have a real problem on our hands that Cisco had better fix, pronto. I would appreciate reports of others as to similar issues or resolutions which they found effective. JT ___ 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] FAX, IAX and *....Maybe I'm dreaming...:-)
How is Vonage doing it? http://www.vonage.com/features_fax.php John Breeden Hawaii -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Underwood Sent: Sunday, December 14, 2003 2:24 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FAX, IAX and *Maybe I'm dreaming...:-) ProvoCityPower wrote: The question asked here, why on earth you want to push fax data over a VoIP link at all. Fax compression isn't very efficient. may speak volumes about the future role of VOIP. My plans are to role out a VOIP connection to thousands of Customers. Many have legacy fax equipment. Am I to assume that they will toss out their fax equipment and join the PC based faxing crowd? I don't think I can control this. If I am going to offer an aternative to the legacy wire providers then I have to offer a comparable service. One that for example allows a customer to use a legacy fax machine in the same way. If this thought sidetracks the intent of this thread, you have my apologies, but I do think that legacy fax functionality is essential. Legacy FAX will be very important for years. The last people to abandon it are the most senior managers. Therefore, apart from anything else, their buyin to a change to VoIP depends on keeping their olde worlde FAX facilities alive. However, I think this has nothing to do with the original poster's intent. Sending FAXes over IP as audio is dumb. Its troublesome, error prone, and consumes too much bandwidth. The right way is to use a FAX modem to translate between audio and the digital data stream of the image itself. Then the compact image data can be conveyed reliably. If the far end is not using IP, another modem can be used to return the FAX to its analogue form for delivery. If you wish to interwork traditional analogue FAX machines, with newer IP capable FAX machines you *have* to do this. This is what the H323 protocols do. T.38 is the relevant FAX over IP protocol. I don't know if there is a similar SIP standard, but there should be. FAX to e-mail and e-mail to FAX is the other inportant way to blend the old with the new. Regards, Steve ___ 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] Cisco 7960 lockups - any experiences?
Hello John, I have a similar story with one of mine.. If i use a cell phone near it, it reboots :) The other ones are fine.. I think it's a hardware problem, but it only reveals itself with the cell phone radiation : Sunday, December 14, 2003, 10:59:14 PM, you wrote: JT This is almost certainly not an Asterisk-specific posting, but due to JT my inability to find a VoIP-focused Cisco list, I'll post here in the JT hopes of finding a more diverse user community. JT I am using a Cisco 7960 (version 6.0 SIP firmware) with Asterisk, and JT have been experiencing situations where the phone locks up. Locks JT up means that the bottom part of the screen (Your current options JT and the redial/newcall/cfwdall keys) disappears, and all keys on the JT keypad are non-functional except for the *-6-settings reboot keys. JT Previous software (4.4) exhibited the same symptoms. I replaced the JT 7960 with a brand new 7960G, thinking it was bad hardware. Same JT symptoms. I replaced the power supply, thinking that perhaps could JT be the problem Same symptoms. I upgraded to 6.0. Same symptoms. I JT replaced the ethernet cables. Same symptoms. I invoked an ancient JT curse-removal spell involving chicken bones and eye of newt over the JT phone. Same symptoms. A phone elsewhere in the same office (same JT switch, etc) works with no problems, but does not have a PC hooked to JT the 'PC' ethernet port. In fact, the first 7960 I tried, with the JT swapped out power supply, works well elsewhere. JT I have used 79xx boxes in many circumstances, and all have performed JT admirably. However, I rarely have configured them so that a PC is JT connected to the other side of the device. The PC is running Windows JT XP, and sees mild SSH/email/web traffic. I cannot say if there is a JT correlation between traffic volume and failure intervals; I have no JT data. JT The only thing that remains the same is the PC on one side, and the JT switch port on the other. Previously, the PC attached to that switch JT port had worked without any problems for quite some time before I put JT the 7960 in place. Even if the problem is the PC or the switchport, JT a failure or malfunction on one of those two components should not JT cause the phone to malfunction. If it is a problem with the 7960 JT pushing traffic across the tiny built-in switch, then we have a real JT problem on our hands that Cisco had better fix, pronto. JT I would appreciate reports of others as to similar issues or JT resolutions which they found effective. JT JT -- Best regards, Nunomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Crash System Command
When I run the asterisk System command my asterisk crashes. When I monitor the console this is the error I get. Any suggestions? exten = 1,1,System(ls) this is the error I get: [EMAIL PROTECTED] asterisk]# Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Crash System Command
Using latest cvs? What distro? because it doesn't happen to mine. bkw On Sun, 14 Dec 2003, Kevin wrote: When I run the asterisk System command my asterisk crashes. When I monitor the console this is the error I get. Any suggestions? exten = 1,1,System(ls) this is the error I get: [EMAIL PROTECTED] asterisk]# Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe ___ 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 Crash System Command
Thanks for the feedback on this. I just downloaded the latest CVS and I'm not sure why it's doing this. -Original Message- From: Brian West [mailto:[EMAIL PROTECTED] Sent: Sunday, December 14, 2003 8:55 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk Crash System Command Using latest cvs? What distro? because it doesn't happen to mine. bkw On Sun, 14 Dec 2003, Kevin wrote: When I run the asterisk System command my asterisk crashes. When I monitor the console this is the error I get. Any suggestions? exten = 1,1,System(ls) this is the error I get: [EMAIL PROTECTED] asterisk]# Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)
Did DVD players have to accommodate VHS tapes? Did VHS players have toaccept beta? Why does VoIP have to deal with an accent protocol that can't handlelossy audio, nor irregular delays?Also why should we be soo wasteful when fax machines need a 80K codec toget the data across IP, and the faster machines I see say 15 secs perpage. So why should we send 1.2meg when 150k is fine? Also who says Fax should ever be required on IP? My office has beenusing VoIP for all voice traffic for over a year now, but always leftthe fax machine on a analog line. The analog line was cheap enough tonot be a concern. -- Steven Critchfield [EMAIL PROTECTED] I'm fairly new here and don't mean to be contentious. We all have different perspectives as to what VOIP should be. My goal is to replace analog lines, not supplement them. I'm talking residential installations. I don't think I can ask these folks to leave their fax on an anolog line? I think that if we start deciding things for the Customer, then VOIP will be seen as an elitist toy for digitally inclined, instead of an acceptable alternative for the masses. No offense to the anti-fax coalition. Jeff
Re: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)
Hi Jeff, I live in Provo and I think I understand the application you're referring to. Some folks in my neighborhood have been getting to be the beta testers for these cool new fiber links that the city is supposed to be laying out. If I only lived a few blocks over, I would be able to get one too. Darn. Anyway, I've been following this thread, and I'm wondering if an alternative might be to provide some sort of fax jack on the hardware you provide the customer that your network could notice and then treat differently from regular voice data? An even better alternative would be if asterisk could recognize a fax machine on the end of the line and use a different protocol or codec that would work with faxes. It sounds like some of the contributors to this thread were saying this is possible. But I'm not sure--I'm pretty new to asterisk and VoIP in general, so I could be wrong. Carl ProvoCityPower wrote: Did DVD players have to accommodate VHS tapes? Did VHS players have to accept beta? Why does VoIP have to deal with an accent protocol that can't handle lossy audio, nor irregular delays? Also why should we be soo wasteful when fax machines need a 80K codec to get the data across IP, and the faster machines I see say 15 secs per page. So why should we send 1.2meg when 150k is fine? Also who says Fax should ever be required on IP? My office has been using VoIP for all voice traffic for over a year now, but always left the fax machine on a analog line. The analog line was cheap enough to not be a concern. -- Steven Critchfield [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] I'm fairly new here and don't mean to be contentious. We all have different perspectives as to what VOIP should be. My goal is to replace analog lines, not supplement them. I'm talking residential installations. I don't think I can ask these folks to leave their fax on an anolog line? I think that if we start deciding things for the Customer, then VOIP will be seen as an elitist toy for digitally inclined, instead of an acceptable alternative for the masses. No offense to the anti-fax coalition. Jeff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Gateway Integration
yes. Cisco 2612 Router with 2 x FXO's and 2 x FXS's. Works well using H323, and gnugk. Steve. Bruce Hedreen [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 15/12/2003 09:57 AM Please respond to asterisk-users To: [EMAIL PROTECTED] cc: Subject: [Asterisk-Users] Cisco Gateway Integration Has anyone succesfully integrated * with a cisco voice gateway ?
RE: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)
It's just my lowly opinion but I too must agree when it comes to the consumer/soho (1 to 3 line) markets. CAUTION!!, DANGER!! Marketing Hat On!! Vonage, the most "visible" marketer of a voip consumer product must also agree. Vontage offers anip "fax line". usingcisco's ata. Vontagemust see some good reason for doing so. (I assume it's h.323). AT, MCI and Time Warner will be competing directly against Vonage when they introducetheir consumer voip products. I'd bet they too will be offering an ip fax line. Odds are you willbe competing against them too. If I was vonage I'd be telling the world how important a ip fax line was :-) Second, the residential/soho market almost demands replacement ofanalog with voip. It's almost impossible to justify the roi unless you do. Marketing Hat Off John Breeden Hawaii I'm fairly new here and don't mean to be contentious. We all have different perspectives as to what VOIP should be. My goal is to replace analog lines, not supplement them. I'm talking residential installations. I don't think I can ask these folks to leave their fax on an anolog line? I think that if we start deciding things for the Customer, then VOIP will be seen as an elitist toy for digitally inclined, instead of an acceptable alternative for the masses. No offense to the anti-fax coalition. Jeff
RE: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)
It's just my lowly opinion but I too must agree when it comes to the consumer/soho (1 to 3 line) markets. CAUTION!!, DANGER!! Marketing Hat On!! Vonage, the most visible marketer of a voip consumer product must also agree. Vontage offers an ip fax line. using cisco's ata. Vontage must see some good reason for doing so. (I assume it's h.323). AT, MCI and Time Warner will be competing directly against Vonage when they introduce their consumer voip products. I'd bet they too will be offering an ip fax line. Odds are you will be competing against them too. If I was vonage I'd be telling the world how important a ip fax line was :-) Personally, I dont think that the world in general really cares about an ip fax line. All they want is a system that works all the time/every time and doesn't require elaborate and convoluted setup. They're not ooohing and ahhhing about oooh, this uses VoIP. They just know that they can stop spending $30/mo on an analog phone line and they get their long distance either flat rate or for an absurdly low per minute rate. Don't sell it as VoIP. Sell it as a total replacement for the analog line. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] can X100P detect phone pick up like an answering machine
If Asterisk is configured as a simple answering machine replacement with the X100P connected to PSTN line. No FXS ports in the Asterisk machine. Standard phones are connect in parallel with the X100P like you would a regular answering machine. Can Asterisk detect that a phone has been picked up and cancel the outgoing message and/or voice recording? What about if the phones are connected to the pass-through port of the X100P? I know some PC software with voice modems can do this, just wondering if X100P/Asterisk can do it? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James Sharp Sent: Sunday, December 14, 2003 6:01 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FAX, IAX and *Maybe I'm dreaming...:-) It's just my lowly opinion but I too must agree when it comes to the consumer/soho (1 to 3 line) markets. CAUTION!!, DANGER!! Marketing Hat On!! Vonage, the most visible marketer of a voip consumer product must also agree. Vontage offers an ip fax line. using cisco's ata. Vontage must see some good reason for doing so. (I assume it's h.323). AT, MCI and Time Warner will be competing directly against Vonage when they introduce their consumer voip products. I'd bet they too will be offering an ip fax line. Odds are you will be competing against them too. If I was vonage I'd be telling the world how important a ip fax line was :-) Personally, I dont think that the world in general really cares about an ip fax line. All they want is a system that works all the time/every time and doesn't require elaborate and convoluted setup. They're not ooohing and ahhhing about oooh, this uses VoIP. They just know that they can stop spending $30/mo on an analog phone line and they get their long distance either flat rate or for an absurdly low per minute rate. Don't sell it as VoIP. Sell it as a total replacement for the analog line. broadband phone ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 lockups - any experiences?
I noticed that on earlier versions of the firmware I was able to crash these phones with a flood ping (ping -f phone-ip). I send this notice to Cisco but never followed up on it. How are you powering this phone, POE or wall cube? -Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nuno Cruz Sent: Sunday, December 14, 2003 8:09 PM To: John Todd Subject: Re: [Asterisk-Users] Cisco 7960 lockups - any experiences? Hello John, I have a similar story with one of mine.. If i use a cell phone near it, it reboots :) The other ones are fine.. I think it's a hardware problem, but it only reveals itself with the cell phone radiation : Sunday, December 14, 2003, 10:59:14 PM, you wrote: JT This is almost certainly not an Asterisk-specific posting, but due JT to my inability to find a VoIP-focused Cisco list, I'll post here in JT the hopes of finding a more diverse user community. JT I am using a Cisco 7960 (version 6.0 SIP firmware) with Asterisk, JT and have been experiencing situations where the phone locks up. JT Locks up means that the bottom part of the screen (Your current options JT and the redial/newcall/cfwdall keys) disappears, and all keys on the JT keypad are non-functional except for the *-6-settings reboot keys. JT Previous software (4.4) exhibited the same symptoms. I replaced the JT 7960 with a brand new 7960G, thinking it was bad hardware. Same JT symptoms. I replaced the power supply, thinking that perhaps could JT be the problem Same symptoms. I upgraded to 6.0. Same symptoms. JT I replaced the ethernet cables. Same symptoms. I invoked an JT ancient curse-removal spell involving chicken bones and eye of newt JT over the phone. Same symptoms. A phone elsewhere in the same JT office (same switch, etc) works with no problems, but does not have JT a PC hooked to the 'PC' ethernet port. In fact, the first 7960 I JT tried, with the swapped out power supply, works well elsewhere. JT I have used 79xx boxes in many circumstances, and all have performed JT admirably. However, I rarely have configured them so that a PC is JT connected to the other side of the device. The PC is running JT Windows XP, and sees mild SSH/email/web traffic. I cannot say if JT there is a correlation between traffic volume and failure intervals; JT I have no data. JT The only thing that remains the same is the PC on one side, and the JT switch port on the other. Previously, the PC attached to that JT switch port had worked without any problems for quite some time JT before I put the 7960 in place. Even if the problem is the PC or JT the switchport, a failure or malfunction on one of those two JT components should not cause the phone to malfunction. If it is a JT problem with the 7960 pushing traffic across the tiny built-in JT switch, then we have a real problem on our hands that Cisco had better fix, pronto. JT I would appreciate reports of others as to similar issues or JT resolutions which they found effective. JT JT -- Best regards, Nunomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MeetMe: Zap channels don't ever disconnect. . .
I was playing around with conferencing tonight. I was able to place a bunch of SIP phones and a couple of my Zap FXS phones into a conference. So I thought, Let's see what it's like when people come in from outside. So I called a friend and had him call in on one of my Zap channels, WHICH IS CONNECTED TO MY POTS LINE THAT DOESN'T DO DISCONNECT SUPERVISION. When he hung up the phone stayed in the conference room until I manually destroyed the channel. I had him call back and come into the conference again; this time we tried a # tone at the moment he left, but that didn't do any good either. I waited perhaps a half-hour to see if *any* kind of timer would release the line, but no dice. Anyone know of a way of doing this when the scumbag ILEC won't give you supervision? Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MeetMe: Zap channels don't ever disconnect. . .
On Mon, 15 Dec 2003, Brian Capouch wrote: Anyone know of a way of doing this when the scumbag ILEC won't give you supervision? Probably not much. Try turning on callprogress and/or busydetect - it MIGHT help. But the only way to do this right is with supervision on an analog trunk or with digital circuits. -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)
Hi Steve, - Original Message - ?From: Steve Underwood [EMAIL PROTECTED] ... . However, I think this has nothing to do with the original poster's intent. You're right. Let's go back to my original problem. There is any chance to make RxFax work with any type of fax machine? It works for me just with a single (hardware) one. The TxFax works with all of them. ..then... we can think further, to support fax data (not fax audio) over slow links..:-) Thank you and best regards, Dan P.S. I think that RxFax and TxFax applications, together with the fax suport in * (tone detection and routing) has a huge potential... keep up the good work. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
