RE: [Asterisk-Users] Application Faxing using SIP
Sorry, I didn't intend to imply I was sharing the server. It's a root server and I control everything on it. The only thing running on it is my application - it's not shared with anyone or anything else.Technical Support [EMAIL PROTECTED] wrote: If you are sharing a box at an ASP, you might have just identified the cause of your problems. Faxing is very time sensitive. With voice, you won't notice or care if there are brief dropouts of audio. With fax, these will cause resend of the raster line (hence the long delays). If your box is shared with other apps, you may not be getting the time slices you need (very different from overall CPU power you are getting).Can you get onto your own box at the ASP?MD From: J Poz [mailto:[EMAIL PROTECTED] Sent: Saturday, February 18, 2006 11:35 PMTo: Technical Support; Asterisk Users Mailing List - Non-Commercial Discussion; Philip EdelbrockSubject: RE: [Asterisk-Users] Application Faxing using SIPMD,Using an analog line is not an option for my service. My application runs on a ROOT SERVER of an ASP. So I can do anything I want to the server but I can't connect to or get external analog lines. So my options are doing faxing via the Internet (VOIP/SIP) or use a faxing service. But my experience with faxing services has not been too good as I've mentioned.Does your company provide an affordable, reliable, and somewhat real-time faxing service? Or can you recommend one? Otherwise, I have to experiment and try to see the results I can get with doing Internet faxing. Remember my experience so far with fax service providers - single faxes take 40+ minutes to eventually be sent (and the delays are within the faxing service and and not the receiving fax line - I've researched this). Technical Support [EMAIL PROTECTED] wrote: J:We developed the mail2fax application (www.generationd.com) - so we should be able to give some insight. I think you are confusing the time to "process" the incoming (by email) fax document, and the time to fax the document. Fax over IP causes an enormous number of retries - thus delays. I would suggest you do some experimenting with an analog line connected to your asterisk box.MD From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J PozSent: Saturday, February 18, 2006 4:17 PMTo: Philip Edelbrock; Asterisk Users Mailing List - Non-Commercial DiscussionCc: J PozSubject: Re: [Asterisk-Users] Application Faxing using SIP Thanks for the information. I prefer to try to develop/configure something myself versus using an external provider. I currently use one (have tried another) and there are no guarantees on reliability, timelines, etc. I'm in need of as close to real-time response as possible (assuming the fa x machineson the other end are operational). Something like within 5 minutes 85% of the time. My experience with 2 external mailtofax providers so far is terrible. Just yesterday, a simple 1 page fax took 42 minutes to finally send it. Other scenarios were 20, 26 minutes, etc. The provider tells me that they don't gaurantee anything but most of the time sending a single page fax within 5 minutes 80% of the time. I wish I would see thatbut so far I'm seeing terrible response times from the few I've tried. I also need to know status of fax (in queue, failed, etc) in real-time so my application and react appropriately (send notification to support staff, etc).I've found one that does have some sort of guarentee by the cost of through the roof and would kill my business model/plan (and the gaurantees are wishy washy). So I think I need to "control" my own destiny.And this definitely is not anything related to spam fax, etc. - legit business but right now can't fully reveal.So I'll have to research abit on IAXModem to use it. But your suggestion is a good one. Can you share what Asterisk configuration you use to both receive the iaxmodem feed and interface to the VOIP provider for such a configuration.Thanks, JPhilip Edelbrock [EMAIL PROTECTED] wrote: On Feb 18, 2006, at 11:35 AM, J Poz wrote: I have a specific business problem that I'm hoping someone has ideas and/or has already worked out a solution. My application needs to be able to automatically create and issue faxes to many different fax machines. The volume is going to be very high. And it is only about sending faxes and not receiving them. ; My application is hosted by an ASP but the Linux (Fedora 2) server is mine (dedicated). So the option of having PSTN lines to do faxes is not an option since I don't own nor can put anything in the data center. I found a SIP/VOIP provider that says they do faxing (and I can connect to them using my own device (meaning asterisk or something else if necessary)). Their requirement for faxing to work on their end is to make sure i send them via their voip service using G.711 codec. So I've done alot of research on faxing and asterisk and hyl
[Asterisk-Users] Application Faxing using SIP
I have a specific business problem that I'm hoping someone has ideas and/or has already worked out a solution.My application needs to be able to automatically create and issue faxes to many different fax machines. The volume is going to be very high. And it is only about sending faxes and not receiving them.My application is hosted by an ASP but the Linux (Fedora 2) server is mine (dedicated). So the option of having PSTN lines to do faxes is not an option since I don't own nor can put anything in the data center. I found a SIP/VOIP provider that says they do faxing (and I can connect to them using my own device (meaning asterisk or something else if necessary)). Their requirement for faxing to work on their end is to make sure i send them via their voip service using G.711 codec.So I've done alot of research on faxing and asterisk and hylafax but I' m still at a loss. For starters, what is the architecture that I need?my application -- QUESTION MARK??? VOIP Provider --- PSTN --- Fax Machine.So first question, what should QUESTION MARK be? Is it just Asterisk or a combination of Asterisk and something like hylafax (fax manager). And depending on that answer, what is the configuration that has to be made on it. Even reference to material that explains the configuration would be very helpful to me at this time. Thanks in advance for the help,J... What are the most popular cars? Find out at Yahoo! Autos ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Application Faxing using SIP
Thanks for the information. I prefer to try to develop/configure something myself versus using an external provider. I currently use one (have tried another) and there are no guarantees on reliability, timelines, etc. I'm in need of as close to real-time response as possible (assuming the fax machineson the other end are operational). Something like within 5 minutes 85% of the time. My experience with 2 external mailtofax providers so far is terrible. Just yesterday, a simple 1 page fax took 42 minutes to finally send it. Other scenarios were 20, 26 minutes, etc. The provider tells me that they don't gaurantee anything but most of the time sending a single page fax within 5 minutes 80% of the time. I wish I would see thatbut so far I'm seeing terrible response times from the few I've tried. I also need to know status of fax (in queue, failed, etc) in real-time so my application and react appropriately (send notification to support staff, etc).I've found one that does have some sort of guarentee by the cost of through the roof and would kill my business model/plan (and the gaurantees are wishy washy). So I think I need to "control" my own destiny.And this definitely is not anything related to spam fax, etc. - legit business but right now can't fully reveal.So I'll have to research abit on IAXModem to use it. But your suggestion is a good one. Can you share what Asterisk configuration you use to both receive the iaxmodem feed and interface to the VOIP provider for such a configuration.Thanks, JPhilip Edelbrock [EMAIL PROTECTED] wrote: On Feb 18, 2006, at 11:35 AM, J Poz wrote: I have a specific business problem that I'm hoping someone has ideas and/or has already worked out a solution. My application needs to be able to automatically create and issue faxes to many different fax machines. The volume is going to be very high. And it is only about sending faxes and not receiving them. My application is hosted by an ASP but the Linux (Fedora 2) server is mine (dedicated). So the option of having PSTN lines to do faxes is not an option since I don't own nor can put anything in the data center. I found a SIP/VOIP provider that says they do faxing (and I can connect to them using my own device (meaning asterisk or something else if necessary)). Their requirement for faxing to work on their end is to make sure i send them via their voip service using G.711 codec. So I've done alot of research on faxing and asterisk and hylafax but I' m still at a loss. F or starters, what is the architecture that I need? my application -- QUESTION MARK??? VOIP Provider --- PSTN --- Fax Machine. So first question, what should QUESTION MARK be? Is it just Asterisk or a combination of Asterisk and something like hylafax (fax manager). And depending on that answer, what is the configuration that has to be made on it. Even reference to material that explains the configuration would be very helpful to me at this time. Thanks in advance for the help,The missing link might be iaxmodem. It has two interfaces: IAX channel for asterisk, and a serial device (in /dev/) which emulates a faxmodem. Then, fax away using hylafax. I have tried faxing over SIP through a provider (broadvoice) to a coworker's fax on the pstn this way, and it worked. I haven't done any testing in volume, though.So you would have something like:Doc - hylafax - iaxmodem - * - voip provider - pstn - fax machinePhilPS- I suppose if you had multiple SIP accounts with a provider, you could create multiple iaxmodems and do things in parallel (assuming enough bandwidth and cpu).PPS- I hope you're not doing fax-spamming with this set up! ;') Yahoo! Autos. Looking for a sweet ride? Get pricing, reviews, & more on new and used cars.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Application Faxing using SIP
MD,Using an analog line is not an option for my service. My application runs on a ROOT SERVER of an ASP. So I can do anything I want to the server but I can't connect to or get external analog lines. So my options are doing faxing via the Internet (VOIP/SIP) or use a faxing service. But my experience with faxing services has not been too good as I've mentioned.Does your company provide an affordable, reliable, and somewhat real-time faxing service? Or can you recommend one? Otherwise, I have to experiment and try to see the results I can get with doing Internet faxing. Remember my experience so far with fax service providers - single faxes take 40+ minutes to eventually be sent (and the delays are within the faxing service and and not the receiving fax line - I've researched this). Technical Support [EMAIL PROTECTED] wrote: J:We developed the mail2fax application (www.generationd.com) - so we should be able to give some insight. I think you are confusing the time to "process" the incoming (by email) fax document, and the time to fax the document. Fax over IP causes an enormous number of retries - thus delays. I would suggest you do some experimenting with an analog line connected to your asterisk box.MD From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J PozSent: Saturday, February 18, 2006 4:17 PMTo: Philip Edelbrock; Asterisk Users Mailing List - Non-Commercial DiscussionCc: J PozSubject: Re: [Asterisk-Users] Application Faxing using SIP Thanks for the information. I prefer to try to develop/configure something myself versus using an external provider. I currently use one (have tried another) and there are no guarantees on reliability, timelines, etc. I'm in need of as close to real-time response a s possible (assuming the fax machineson the other end are operational). Something like within 5 minutes 85% of the time. My experience with 2 external mailtofax providers so far is terrible. Just yesterday, a simple 1 page fax took 42 minutes to finally send it. Other scenarios were 20, 26 minutes, etc. The provider tells me that they don't gaurantee anything but most of the time sending a single page fax within 5 minutes 80% of the time. I wish I would see thatbut so far I'm seeing terrible response times from the few I've tried. I also need to know status of fax (in queue, failed, etc) in real-time so my application and react appropriately (send notification to support staff, etc).I've found one that does have some sort of guarentee by the cost of through the roof and would kill my business model/plan (and the gaurantees are wishy washy). So I think I need to "control" my own destiny.And this definit ely is not anything related to spam fax, etc. - legit business but right now can't fully reveal.So I'll have to research abit on IAXModem to use it. But your suggestion is a good one. Can you share what Asterisk configuration you use to both receive the iaxmodem feed and interface to the VOIP provider for such a configuration.Thanks, JPhilip Edelbrock [EMAIL PROTECTED] wrote: On Feb 18, 2006, at 11:35 AM, J Poz wrote: I have a specific business problem that I'm hoping someone has ideas and/or has already worked out a solution. My application needs to be able to automatically create and issue faxes to many different fax machines. The volume is going to be very high. And it is only about sending faxes and not r eceiving them. My application is hosted by an ASP but the Linux (Fedora 2) server is mine (dedicated). So the option of having PSTN lines to do faxes is not an option since I don't own nor can put anything in the data center. I found a SIP/VOIP provider that says they do faxing (and I can connect to them using my own device (meaning asterisk or something else if necessary)). Their requirement for faxing to work on their end is to make sure i send them via their voip service using G.711 codec. So I've done alot of research on faxing and asterisk and hylafax but I' m still at a loss. F or starters, what is the architecture that I need? my application -- QUESTION MARK??? VOIP Provider --- PSTN --- Fax Machine. So first question, what should QUESTION MARK be? Is it just Asterisk or a combination of Asterisk and something like hylafax (fax manager). And depending on that answer, what is the configuration that has to be made on it. Even reference to material that explains the configuration would be very helpful to me at this time. Thanks in advance for the help,The missing link might be iaxmodem. It has two interfaces: IAX channel for asterisk, and a serial device (in /dev/) which emulates a faxmodem. Then, fax away using hylafax. I have tried faxing over SIP through a provider (broadvoice) to a coworker's fax on the pstn this way, and it worked. I haven't done any testing in volume, though.So you would have something like:Doc - hylafax - iaxmodem - * - voip provider - pstn - fax machinePhil
RE: [Asterisk-Users] How does Norvergence do it ?
Wow, that was GREAT info... I'm in the NY tri-state area so I'm sure I will run into them as competition. Thanks for sending that link - it's unbelievable; also proves the power of such emailing lists where we can share this type of information about unethical companies!Neil Grant [EMAIL PROTECTED] wrote: Found this:http://w3.ripoffreport.com/reports/ripoff89155.htmMany other nasty stories about them too.--Cheers,Neil-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: 04 May 2004 12:51To: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] How does Norvergence do it ?So a guy shows up at the the office, after making anappointment with the office manager / receptionist to talk'phone systems'.After her eyes glaze over, with him talking T1 andFrame-Relay I get to see him. He's from Norvergence. Welldressed. Tells me they can do a T1 for $79, with unlimitedlocal long distance for free. It also does 'internet'.'Just give me copies of your phone bill'. I ask somequestions, like number porting, like provisioning of DIDnumbers, like CIR on the data etc. Now HIS eyes glaze over.That's technical talk ... He's just there to follow up onthe appointment and 'qualify' the customer to see if we areworthy of their cheap service. After I looked at theirwebsite, I can hear 'quack quack'.Cheers,WW Willy WoutersypOne Publishing___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!?Win a $20,000 Career Makeover at Yahoo! HotJobs
RE: [Asterisk-Users] How does Norvergence do it ?
Yes, I would like to see the contract.Michael Miller [EMAIL PROTECTED] wrote: I have a good friend who used to be a sales rep for them. The entire sales pitch is based on making the customer believe that they are lucky to have been offered the opportunity to beome a Norvergence customer as they are extremely selective. If any technical question where to come up, he was trained to let them know that they where not selected to be part of the program and to move on quickly. I do have a copy of their contract if anyone is interested is taking a look at it. It is an interesting piece of legal work. Michael From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J PozSent: Tuesday, May 04, 2004 9:45 AMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] How does Norvergence do it ? Wow, that was GREAT info... I'm in the NY tri-state area so I'm sure I will run into them as competition. Thanks for sending that link - it's unbelievable; also proves the power of such emailing lists where we can share this type of information about unethical companies!Neil Grant [EMAIL PROTECTED] wrote: Found this:http://w3.ripoffreport.com/reports/ripoff89155.htmMany other nasty stories about them too.--Cheers,Neil-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: 04 May 2004 12:51To: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] How does Norvergence do it ?So a guy shows up at the the office, after making anappointment with the office manager / receptionist to talk'phone systems'.After her eyes glaze over, with him talking T1 andFrame-Relay I get to see him. He's from Norvergence. Welldressed. Tells me they can do a T1 for $79, with unlimitedlocal long distance for free. It also does 'internet'.'Just give me copies of your phone bill'. I ask somequestions, like number porting, like provisioning of DIDnumbers, like CIR on the data etc. Now HIS eyes glaze over.That's technical talk ... He's just there to follow up onthe appointment and 'qualify' the customer to see if we areworthy of their cheap service. After I looked at theirwebsite, I can hear 'quack quack'.Cheers,WW Willy WoutersypOne Publishing___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!?Win a $20,000 Career Makeover at Yahoo! HotJobs Do you Yahoo!?Win a $20,000 Career Makeover at Yahoo! HotJobs
[Asterisk-Users] Simple SIP X-Lite Configuration Failing
I keep getting the following Auto-congesting message whenever I try to dial from an X-Lite SIP phone to another one within my LAN. It's a real basic configuration but I am unable to figure out what is happening:
localhost*CLI -- Executing Dial("SIP/jay-de1b", "SIP/jtest|20|tr") in new stack -- Called jtestMay 2 11:47:58 NOTICE[1133742896]: chan_sip.c:1019 auto_congest: Auto-congesting SIP/jtest-6a1e -- SIP/jtest-6a1e is circuit-busy == Everyone is busy at this timeMay 2 11:47:58 WARNING[1226062640]: pbx.c:1198 pbx_extension_helper: No application 'DialCongestion' for extension (sip, 22, 2) == Spawn extension (sip, 22, 2) exited non-zero on 'SIP/jay-de1b'
My setup is very simple and basic:
SIP.conf
[general]port = 5060bindaddr = 0.0.0.0context = sip; Default
[jay]type=friendsecret=jaysipauth=md5nat=yeshost=dynamicreinvite=nocanreinvite=noqualify=100dtmfmode=inbandcallerid="Jay 400"disallow=allallow=gsmcontext=sip
[jtest]type=friendsecret=jaytestsipauth=md5nat=yeshost=dynamicreinvite=nocanreinvite=noqualify=100dtmfmode=inbandcallerid="Jay Test 410disallow=allallow=gsmcontext=sip
extensions.conf
[sip] ; context for X-Lite Clientsexten =11,1,Dial(SIP/jay,20,tr)exten =11,2,Congestionexten =22,1,Dial(SIP/jtest,20,tr)exten =22,2,DialCongestion
Lastly, here's my client setup
Display Name: JayUser Name Authorization User: jay
Password:jaysip Domain/Realm:192.168.1.20 SIP Proxy:192.168.1.20
Display Name: Jay TestUser Name Authorization User: jtest
Password:jaytestsip Domain/Realm:192.168.1.20 SIP Proxy:192.168.1.20
Any help anyone can give me would be appreciated since I've already spentHOURS on this and havemade absolutely no progress in debugging this (didn't find anything in any of the archives nor wiki pages).
J...
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Re: [Asterisk-Users] Simple SIP X-Lite Configuration Failing
Jeremy, I'm missing something hereI understand the congestion part - no problem, I can take that out of extensions.conf.. But I need help in the real problem which is the Auto-congesting and circuit-busy part. Why is it saying the circuit is busy? If siphone "jay" dials "jaytest" it says circuit-busy eventhough jaytest is not busy (and vice-versa). So what am I missing (I know this is simple but I haven't been able to figure it out so why I'm asking for help). J..Jeremy McNamara [EMAIL PROTECTED] wrote: J Poz wrote: -- SIP/jtest-6a1e is circuit-busy == Everyone is busy at this time May 2 11:47:58 WARNING[1226062640]: pbx.c:1198 pbx_extension_helper: No applica tion 'DialCongestion' for extension (sip, 22, 2)Come on... Asterisk is telling you EXACTLY whats wrong... SIP/jtest is circuit-busy and then asterisk cannot find an application named DialCongetsion.Fix your extensions.confJeremy McNamara___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!?Win a $20,000 Career Makeover at Yahoo! HotJobs
[Asterisk-Users] Re: Simple SIP X-Lite Configuration Failing
Can anyone help. I've changed the extensions.conf file as follows:
extensions.conf
[sip] ; context for X-Lite Clientsexten =11,1,Dial(SIP/jay,20,tr)exten =22,1,Dial(SIP/jtest,20,tr)
I'm still getting the Auto-congesting error (and circuit-busy). Does anyone know what is causing this in such a simple configuration?
localhost*CLI -- Executing Dial("SIP/400-c3de", "SIP/410|20|tr") in new stack -- Called 410May 2 19:15:56 NOTICE[1133742896]: chan_sip.c:1021 auto_congest: Auto-congesting SIP/410-a4a1 -- SIP/410-a4a1 is circuit-busy == Everyone is busy at this timeJ Poz [EMAIL PROTECTED] wrote:
I keep getting the following Auto-congesting message whenever I try to dial from an X-Lite SIP phone to another one within my LAN. It's a real basic configuration but I am unable to figure out what is happening:
localhost*CLI -- Executing Dial("SIP/jay-de1b", "SIP/jtest|20|tr") in new stack -- Called jtestMay 2 11:47:58 NOTICE[1133742896]: chan_sip.c:1019 auto_congest: Auto-congesting SIP/jtest-6a1e -- SIP/jtest-6a1e is circuit-busy == Everyone is busy at this timeMay 2 11:47:58 WARNING[1226062640]: pbx.c:1198 pbx_extension_helper: No application 'DialCongestion' for extension (sip, 22, 2) == Spawn extension (sip, 22, 2) exited non-zero on 'SIP/jay-de1b'
My setup is very simple and basic:
SIP.conf
[general]port = 5060bindaddr = 0.0.0.0context = sip; Default
[jay]type=friendsecret=jaysipauth=md5nat=yeshost=dynamicreinvite=nocanreinvite=noqualify=100dtmfmode=inbandcallerid="Jay 400"disallow=allallow=gsmcontext=sip
[jtest]type=friendsecret=jaytestsipauth=md5nat=yeshost=dynamicreinvite=nocanreinvite=noqualify=100dtmfmode=inbandcallerid="Jay Test 410disallow=allallow=gsmcontext=sip
extensions.conf
[sip] ; context for X-Lite Clientsexten =11,1,Dial(SIP/jay,20,tr)exten =11,2,Congestionexten =22,1,Dial(SIP/jtest,20,tr)exten =22,2,DialCongestion
Lastly, here's my client setup
Display Name: JayUser Name Authorization User: jay
Password:jaysip Domain/Realm:192.168.1.20 SIP Proxy:192.168.1.20
Display Name: Jay TestUser Name Authorization User: jtest
Password:jaytestsip Domain/Realm:192.168.1.20 SIP Proxy:192.168.1.20
Any help anyone can give me would be appreciated since I've already spentHOURS on this and havemade absolutely no progress in debugging this (didn't find anything in any of the archives nor wiki pages).
J...
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