Re: [asterisk-users] Hardware that can ring my phone?
On 8/1/07, Linux Lover [EMAIL PROTECTED] wrote: This SOHO PBX box won't interop with Asterisk because it doesn't speak any of the protocols that Asterisk does. This box I tend agree with your evaluation. Still, I was thinking that since all these el-cheapo SOHO PBX boxes support manual attendant call transfer, what's to prevent Asterisk from mimicking an attendant by sending proper DTMF signals and make this box transfer the call to the single analog phone in the business? That is, Asterisk will connect (via RJ-11) to the unit as the attendant's phone, and my real phone (only one in the system) will connect via a second RJ-11 (there could be 4 of them). Or is Asterisk not capable of sending DTMF signals over an RJ-11 connection? You can send arbitrary DTMF over any of Asterisk's channels from the dialplan. I just figured that this level of integration was a bit deeper than you were looking for as a first project. It would be an interesting experiment, to be sure. The biggest issue I'd think would be feedback - you can send the DTMF along the wire, but how do you know that the SOHO box interpreted it correctly? If the only feedback is designed for a human (i.e. auditory), then interpreting those cues with Asterisk would be non-trivial. Do I undestand correctly that with this solution, I will still be able to connect to my analog Verizon phone line with the SIP phone? That is, the outside world will see my phone as an ordinary phone, when in fact I am using a SIP phone? If so, that means that Asterisk does all the magic behind the scene, right? Yes, your Verizon POTS line would go into a FXO port in your server (which in Asterisk would be referenced as the channel Zap/1 - zaptel being Asterisk's TDM driver) and your SIP phone would connect via your standard office network and be referenced as SIP/whateverusernameyouwant. A very simplistic example of bridging a call would be: [from-verizon] exten = s,1,Dial(SIP/whateverusername) Assuming that you'd configured zaptel to route calls that come in on the FXO port to the Asterisk context named from-verizon, then any such calls would immediately cause Asterisk to ring your SIP phone, and if answered to bridge the two calls together. A more complex example that makes them press one to call you and otherwise lets them leave a message: [from-verizon] exten = s,1,Background(Press1ToTalkOr2ToLeaveAMessage) exten = s,n,WaitExten(10) ; timeout exten = t,1,Goto(vm,1) ; invalid exten = i,1,Goto(vm,1) ; press 1 exten = 1,1,Dial(SIP/101,20) exten = 1,n,Goto(vm,1) ; press 2 exten = 2,1,Goto(vm,1) ; all voicemail activity ends up here exten = vm,1,VoiceMail(u101) exten = vm,n,Hangup [from-officephone] exten = *98,1,VoiceMailMain extne = *98,n,Hangup Assuming you've now set up your SIP phone as extension 101, this would play a sound file saying press 1 to talk to 2 to leave a message. If they press 1, your SIP phone rings. If they press 2, they go to voicemail. If they wait 10 seconds without pressing anything, or press something other than 1 or 2, they also go to voicemail. If they press 1 to dial your phone and you don't pick up after 20 seconds, they go to voicemail. On your deskphone (could just as easily be a SIP softphone if you prefer), you can dial *98 to log in and pick up your new voicemail messages. Hope that demystifies some of what you're trying to do. -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem in making SIP call after compiling Asterisk server
Prathap, That response is not sent by Asterisk. What you are most likely getting this from is a packet capture, and what you are referring to is an ICMP message sent as a backward notification by an intermediate router or host. Basically, it sounds like the SIP UDP port (5060) on the Asterisk server (whatever its pingability) is firewalled off or access lists are stopping the traffic, either on a router somewhere in the path or on the Asterisk server itself. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem in making SIP call after compiling Asterisk server
Hi There, I have installed an Asterisk server on Fedora Core, I can able to run the Asterisk Server successfully. But the problem is, my softphone(Xlite) is not getting registered with Asterisk server. From softphone Register request were sent and the Asterisk respond with Destination Unreachable (Host Administratively prohibited) But I can able to ping the Asterisk server from a PC(Where softphone is residing) and vice versa. Please give me your suggestions to make a SIP call successfully. Thanks in advance. Regards, Prathap DISCLAIMER: --- The contents of this e-mail and any attachment(s) are confidential and intended for the named recipient(s) only. It shall not attach any liability on the originator or HCL or its affiliates. Any views or opinions presented in this email are solely those of the author and may not necessarily reflect the opinions of HCL or its affiliates. Any form of reproduction, dissemination, copying, disclosure, modification, distribution and / or publication of this message without the prior written consent of the author of this e-mail is strictly prohibited. If you have received this email in error please delete it and notify the sender immediately. Before opening any mail and attachments please check them for viruses and defect. --- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slightly OT: SNOM PoE
I am using the netgear switch 24 ports and 8 ports w/ snom 360 in a 10/100 network w/ no problem. the actual version of firmware of SNOM is 6.5.10 but the phone works w/ previews version. Look in your network. Bruno. Anthony Cennami wrote: Hello All, I apologize for the slightly off-topic question, but I'm sure that the people best acquainted with the issue would be hanging around here. We recently deployed several Linksys POE switches for some smaller customers (10-24 station) and appear to be suffering from intermittent lock-ups of the SNOM phones attached. Obviously we are running Asterisk for the gateway, but I was curious if anybody has experienced similar issues. Phones will run fine, and then intermittently (and at different times for different ports) the phones will lockup and require a hard reboot. I've read on voip-info that the SNOM phones are apparently sensitive to lower-end network equipment, presumably with PoE only aggravating the problem. Question is, what are people using today to deploy PoE, and more importantly, PoE to SNOM phones? I believe the model we're working with is the SR224P from Linksys, and the entire model line of SNOM (3XX) Could anybody recommend some well-used/tested PoE equipment that you've found successful in your SNOM envionment? Looking for density of 24-ports plus, and ideally some lower end and higher end equipment, to satisfy the needs of the wide variety of customers we do business with? Thanks, anthony -- Anthony Cennami ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruno De Luca, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02 997663.12, Fax: 02 91390172 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware that can ring my phone?
James FitzGibbon wrote: On 8/1/07, *Linux Lover* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: This SOHO PBX box won't interop with Asterisk because it doesn't speak any of the protocols that Asterisk does. This box I tend agree with your evaluation. Still, I was thinking that since all these el-cheapo SOHO PBX boxes support manual attendant call transfer, what's to prevent Asterisk from mimicking an attendant by sending proper DTMF signals and make this box transfer the call to the single analog phone in the business? That is, Asterisk will connect (via RJ-11) to the unit as the attendant's phone, and my real phone (only one in the system) will connect via a second RJ-11 (there could be 4 of them). Or is Asterisk not capable of sending DTMF signals over an RJ-11 connection? You can send arbitrary DTMF over any of Asterisk's channels from the dialplan. I just figured that this level of integration was a bit deeper than you were looking for as a first project. It would be an interesting experiment, to be sure. The biggest issue I'd think would be feedback - you can send the DTMF along the wire, but how do you know that the SOHO box interpreted it correctly? If the only feedback is designed for a human ( i.e. auditory), then interpreting those cues with Asterisk would be non-trivial. Do I undestand correctly that with this solution, I will still be able to connect to my analog Verizon phone line with the SIP phone? That is, the outside world will see my phone as an ordinary phone, when in fact I am using a SIP phone? If so, that means that Asterisk does all the magic behind the scene, right? Yes, your Verizon POTS line would go into a FXO port in your server (which in Asterisk would be referenced as the channel Zap/1 - zaptel being Asterisk's TDM driver) and your SIP phone would connect via your standard office network and be referenced as SIP/whateverusernameyouwant. A very simplistic example of bridging a call would be: [from-verizon] exten = s,1,Dial(SIP/whateverusername) Assuming that you'd configured zaptel to route calls that come in on the FXO port to the Asterisk context named from-verizon, then any such calls would immediately cause Asterisk to ring your SIP phone, and if answered to bridge the two calls together. A more complex example that makes them press one to call you and otherwise lets them leave a message: [from-verizon] exten = s,1,Background(Press1ToTalkOr2ToLeaveAMessage) exten = s,n,WaitExten(10) ; timeout exten = t,1,Goto(vm,1) ; invalid exten = i,1,Goto(vm,1) ; press 1 exten = 1,1,Dial(SIP/101,20) exten = 1,n,Goto(vm,1) ; press 2 exten = 2,1,Goto(vm,1) ; all voicemail activity ends up here exten = vm,1,VoiceMail(u101) exten = vm,n,Hangup [from-officephone] exten = *98,1,VoiceMailMain extne = *98,n,Hangup Assuming you've now set up your SIP phone as extension 101, this would play a sound file saying press 1 to talk to 2 to leave a message. If they press 1, your SIP phone rings. If they press 2, they go to voicemail. If they wait 10 seconds without pressing anything, or press something other than 1 or 2, they also go to voicemail. If they press 1 to dial your phone and you don't pick up after 20 seconds, they go to voicemail. On your deskphone (could just as easily be a SIP softphone if you prefer), you can dial *98 to log in and pick up your new voicemail messages. Hope that demystifies some of what you're trying to do. -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users the way to have * send dtmf is with the D option, w inserts a half second pause. As an example I have a remote location that needs special 911, so they have a landline that connects to a linksys SPA, it doesnt like being passed the destination number through sip, so O do it this way: exten = 911,1,Dial(SIP/08CCB243-911,,D(w911)) works awesome, it connects, plays back the DTMF, and then passes the audio stream to the caller. Anthony ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple pbxes, multiple domains, same user ids?
Benjamin Jacob wrote: Ouch. And I thought I had an answer to my query. I totaly agree abt the long disclaimer nonsense Schmaltz, but I swear by the powers up there, it's the admins over here at my workplace doing all that nonsensical magic, as the mails go out. I wish i had the freedom to use gmail(just like you), thru the day, and not the office mail servers! Do you have any idea as to how do I get rid of this disclaimer whenever I mail to the Asterisk Users mailing list?? Pray, tell me! Btw, did you happen to read my query, or you straight on jumped to the disclaimer? roving eyes, eh? Any answers anyone , to my query(abt multiple pbxes)? Apologies if I am missing something elementary here. cheerz - Ben. C F wrote: Can you please get rid of your awfull long nonsense disclaimer? On 8/1/07, Benjamin Jacob [EMAIL PROTECTED] wrote: Hello good ppl, A couple of questions for multiple pbxes 1. Is it possible to support multiple pbxes in one Asterisk box(using contexts, etc.)? 2. Can we use the domain field in sip.conf to specify the different domains for sip users, having one domain for each pbx? I just tried registering two xlites, with different domain names (with the same specified in sip.conf). But, Asterisk maintains the registration of the latest registree!! thats really sad for me . Any work around for this one(multiple pbx)? I would be zapped and amazed if multiple pbx isn't possible in Asterisk. Help anyone? cheers - Ben. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users you have to do different sip-ids, I am guessing you are probably using the extension #, you dont need to do that. What do you mean by multiple-pbx's anyway? I hope you don't mean multiple instances of *.What I am sure you mean is multiple dial plans, and yes, * is multi-tenant friendly. What we do for uniqueness is use the last 8 digits of the device mac addr or other unique number followed by a dash - followed by the extension number. Anthony ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - How to switch headphones between softphones on Linux ?
Hello, A little Off-Topic but how can you easily switch microphone and headphone from one softphone to another on a Linux KDE platform ? I use Skype (for incoming calls mainly) and Twinkle (for outgoing and incoming calls). I could't find any practical way to quickly switch audio from one application to another. Ideally, I would like to register some kind of virtual audio resources to each application and then use another software to quickly switch each virtual audio ressource to real audio resources whenether a call comes in. Today, I have to use Twinkle's Audio tab in Edit/Systems settings, for instance, and that takes me too much time. Any suggestion ? Cheers ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple pbxes, multiple domains, same user ids?
Anthony Francis wrote: Hello good ppl, A couple of questions for multiple pbxes 1. Is it possible to support multiple pbxes in one Asterisk box(using contexts, etc.)? 2. Can we use the domain field in sip.conf to specify the different domains for sip users, having one domain for each pbx? I just tried registering two xlites, with different domain names (with the same specified in sip.conf). But, Asterisk maintains the registration of the latest registree!! thats really sad for me . Any work around for this one(multiple pbx)? I would be zapped and amazed if multiple pbx isn't possible in Asterisk. Help anyone? cheers - Ben. you have to do different sip-ids, I am guessing you are probably using the extension #, you dont need to do that. What do you mean by multiple-pbx's anyway? I hope you don't mean multiple instances of *.What I am sure you mean is multiple dial plans, and yes, * is multi-tenant friendly. What we do for uniqueness is use the last 8 digits of the device mac addr or other unique number followed by a dash - followed by the extension number. Anthony Thanks Anthony. I definitely don't mean multiple instances of asterisk. Multiple dial plans, hmm.. yes.. in a way. Multiple pbx ... in short, provide pbxes for two entirely different organizations, say, Microsoft and IBM (can i use these names in here? ;-) ). Each would have many extensions, but each office can have identical extensions, e.g. you can have extensions 4001 in both. But one would be [EMAIL PROTECTED] and the other would be [EMAIL PROTECTED] . [EMAIL PROTECTED] should be able to call any user within Microsoft. To step outside the organization, you would put in some logic(dialplans). So, i want to have pbx for microsoft and another pbx for IBM. Is it possible to have two or more pbxes within one Asterisk instance. Hope you got my point. cheerz - Ben. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting GSM Phone to Asterisk Box
Hi Folks, Thanks for the suggestions so far! Please keep them coming. I plan ot summarize and post it for the record.maybe work it into the faq somewhere, who knows. Jeng Steve Totaro [EMAIL PROTECTED] wrote: Chan_bluetooth is now chan_mobile and included in trunk/asterisk-addons. That would be my suggestion. It works very well. Thanks, Steve Totaro Andrew Joakimsen wrote: On 7/31/07, *Jeng Yu* wrote: Hi All, I have a telephony project for which I need to build a prototype to demo for management. The prototype must work on a GSM phone network. In the demo system, a call from GSM phone comes into the demo box. The demo box runs CallWeaver. Callweaver picks up the GSM call, answers it and plays a sould file, then dials out to a second GSM phone somewhere and connects them so they talk. My question are these: 1. Is this a job for Callweaver/asterisk system? 2. if not, what package out there would handle this? I was reading the Asterisk doc and it mentions that a modem could be used with an Asterisk system, but does not show how it can be configured to function in the system. Does anyone here know how? There are various PCI and Ethernet GSM interfaces that should work with Asterisk. These would be your best choice in terms of reliability and performance for a production system. It wouldn't hurt to have a fixed, semi-directional, antenna pointed to the tower either. I wouldn't recommend for a production system, but you can also use a GSM - Analog adapter such as the CellDock or DockNTalk I think some people call this Tellular and also somewhere is chan_bluetooth which allows you to place calls through a bluetooth mobile. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Yahoo! Answers - Get better answers from someone who knows. Tryit now.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropouts and echo
Tom Lanyon wrote: Hi all, Can I ask that you please keep my personal address in the To: or CC: in this thread as for some reason I'm only getting half of the list emails coming through, and they're not showing up on the digium pipermail archive either. The list archive on http://marc.info seems to have the whole thread though. Have you tried changing the RTP packet size on the phones from .30(default I believe) to .20? The RTP packet size is currently 0.030 which is the default (I wasn't aware we could change it). Would changing to 0.020 help? Why don't you try it? Echo and drop outs usually require trying many different things (especially echo). That is a place to start. Make sure you document your changes and so you can easily roll them back. Finding your actual problem will most likely be a trial and error process, so start trying. Turn OFF CDP on the phones. The phones don't support CDP as far as I can find; I know Linksys is a subdivision of Cisco, but these phones are actually made by Sipura. As for Echo Canceling, that is the job of the device that does VoIP/PSTN gateway functions. As mentioned before, this is SIP - SIP, so the echo isn't introduced by the PSTN. I'll keep experimenting with volume levels and environmental issues to try and fix the echo. Depending on how you define echo, you can try different things with the phones such as pressing mute, stuffing the handset with something to dampen the sound, have both parties speak softly and then try loud. Again, document your results. What kind of switch are you connecting the phones to? I've seen that behaviour with cheap Repotec switches (24+2Gigabit). Just replacing it with a different one fixed the problem. The switch is indeed a poor quality one. My next step is to replace it with something decent and see if it helps. I wasn't sure whether this could be the cause so it's good to have your input. A switch could cause drop outs but I doubt echo so much. While the first thing you should do is have decent quality gear on your network, I would also look at timing issues on your box and change RTP from packet size to 20. Make sure ZAPTEL or ZTDUMMY is loaded, make sure your system is using RTC and no HPET. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slightly OT: SNOM PoE
Hi guys, one of our German distributors (Allnet) has reasonable PoE switches (price/features). They also have a distribution Channel in the US ( http://www.allnet-usa.com). They at least work pretty ok in our environment. I could imagine that the power isn't very clean. Meaning the voltage changes, there might be peaks, etc. This can also come from the power outlet in the wall, and be given through by the PoE switch to the phones. You could try to run the PoE switch with an electronical Backup (UPS) which cleans the power from the power network. Hi Andrew, the GSM lookup is completly new to us, you're the first one reporting this us. I just checked myself and let the hardware guys check your comment on the GSM phone. We weren't able to reproduce the lock up. Neither UMTS, nor GSM (1800MHz and 800MHz) had any effect on the phone on my desk. Please specify this (used phone, location where you're holding the phone?, etc.). Regards Tim On 8/1/07, Andrew Latham [EMAIL PROTECTED] wrote: I use on a regular basis the D-Link line, they work. With the SNOM you will want to set the ignore Ethernet unplug in case the Ethernet switch restarts (like a Netgear 7248 attached to a cheap fiber trans). Keep in mind that holding a GSM phone real close to some of the SNOM phones will cause them to lock up. On 8/1/07, Anthony Cennami [EMAIL PROTECTED] wrote: Hello All, I apologize for the slightly off-topic question, but I'm sure that the people best acquainted with the issue would be hanging around here. We recently deployed several Linksys POE switches for some smaller customers (10-24 station) and appear to be suffering from intermittent lock-ups of the SNOM phones attached. Obviously we are running Asterisk for the gateway, but I was curious if anybody has experienced similar issues. Phones will run fine, and then intermittently (and at different times for different ports) the phones will lockup and require a hard reboot. I've read on voip-info that the SNOM phones are apparently sensitive to lower-end network equipment, presumably with PoE only aggravating the problem. Question is, what are people using today to deploy PoE, and more importantly, PoE to SNOM phones? I believe the model we're working with is the SR224P from Linksys, and the entire model line of SNOM (3XX) Could anybody recommend some well-used/tested PoE equipment that you've found successful in your SNOM envionment? Looking for density of 24-ports plus, and ideally some lower end and higher end equipment, to satisfy the needs of the wide variety of customers we do business with? Thanks, anthony -- Anthony Cennami ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] [EMAIL PROTECTED] */ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- snom technology AG Tim Koehler Partner Manager [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slightly OT: SNOM PoE
Tim I have a batch of 30 that it does not affect. I also have a batch of 12 that it does effect. I like the SNOM phones and I think that I just got a batch missing some shielding or other component. It is _very_ noticeable when it happens. Andrew On 8/2/07, Tim Koehler [EMAIL PROTECTED] wrote: Hi guys, one of our German distributors (Allnet) has reasonable PoE switches (price/features). They also have a distribution Channel in the US (http://www.allnet-usa.com ). They at least work pretty ok in our environment. I could imagine that the power isn't very clean. Meaning the voltage changes, there might be peaks, etc. This can also come from the power outlet in the wall, and be given through by the PoE switch to the phones. You could try to run the PoE switch with an electronical Backup (UPS) which cleans the power from the power network. Hi Andrew, the GSM lookup is completly new to us, you're the first one reporting this us. I just checked myself and let the hardware guys check your comment on the GSM phone. We weren't able to reproduce the lock up. Neither UMTS, nor GSM (1800MHz and 800MHz) had any effect on the phone on my desk. Please specify this (used phone, location where you're holding the phone?, etc.). Regards Tim On 8/1/07, Andrew Latham [EMAIL PROTECTED] wrote: I use on a regular basis the D-Link line, they work. With the SNOM you will want to set the ignore Ethernet unplug in case the Ethernet switch restarts (like a Netgear 7248 attached to a cheap fiber trans). Keep in mind that holding a GSM phone real close to some of the SNOM phones will cause them to lock up. On 8/1/07, Anthony Cennami [EMAIL PROTECTED] wrote: Hello All, I apologize for the slightly off-topic question, but I'm sure that the people best acquainted with the issue would be hanging around here. We recently deployed several Linksys POE switches for some smaller customers (10-24 station) and appear to be suffering from intermittent lock-ups of the SNOM phones attached. Obviously we are running Asterisk for the gateway, but I was curious if anybody has experienced similar issues. Phones will run fine, and then intermittently (and at different times for different ports) the phones will lockup and require a hard reboot. I've read on voip-info that the SNOM phones are apparently sensitive to lower-end network equipment, presumably with PoE only aggravating the problem. Question is, what are people using today to deploy PoE, and more importantly, PoE to SNOM phones? I believe the model we're working with is the SR224P from Linksys, and the entire model line of SNOM (3XX) Could anybody recommend some well-used/tested PoE equipment that you've found successful in your SNOM envionment? Looking for density of 24-ports plus, and ideally some lower end and higher end equipment, to satisfy the needs of the wide variety of customers we do business with? Thanks, anthony -- Anthony Cennami ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] [EMAIL PROTECTED] */ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- snom technology AG Tim Koehler Partner Manager [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] [EMAIL PROTECTED] */ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Blip every 30 seconds?
Strange issue when I record a file from a phone to the asterisk system I get a blip in the recording every 30 seconds. It's a very small blip, but it is there.It seems like it's only if I'm recording, not when I'm playing back that the issue happens. My SATA drives, ETH0, and my Sangoma card are all on seperate IRQs. Any thoughts on what might be causing this and how to stop it? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with rfc2833
I have the following: pri box incoming/outgoing on box 1 connected through SIP to box 2. The box 1 to box 2 has dtmfmode=rfc2833. With this setting calls going out of box2 through box 1 the sendDTMF() mode does not do anything. When I change dtmfmode=info I at least hear the sendDTMF() digits. Why doesnt rfc2833 work? Jerry ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] radius support
hi, how to add radius support to asterisk 1.4.5? i do make menuselect and i do not see any module or option related to radius, pam, authenticacion or similar. any ideas? thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blip every 30 seconds?
Telephone conversations that are being recorded, are supposed to beep periodically, to alert/remind the recorded person that the conversation is being recorded. Perhaps that is what you are hearing? joe a. On 8/2/2007 at 8:47 AM, Matt [EMAIL PROTECTED] wrote: Strange issue when I record a file from a phone to the asterisk system I get a blip in the recording every 30 seconds. It's a very small blip, but it is there.It seems like it's only if I'm recording, not when I'm playing back that the issue happens. My SATA drives, ETH0, and my Sangoma card are all on seperate IRQs. Any thoughts on what might be causing this and how to stop it? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blip every 30 seconds?
Matt wrote: Strange issue when I record a file from a phone to the asterisk system I get a blip in the recording every 30 seconds. It's a very small blip, but it is there.It seems like it's only if I'm recording, not when I'm playing back that the issue happens. My SATA drives, ETH0, and my Sangoma card are all on seperate IRQs. Any thoughts on what might be causing this and how to stop it? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users There is option q to be quiet the Record() application. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE220B
Hi, Has anyone ever had any problem with the TE220B card with it showing up as four ports instead of two. I RMA'd the first one with the retailer (Digium tech advice), but I just got another brand new card and it is coming up as four ports again. The card identifier is showing 0420 when I do lspci. Has this happened to anyone and if so is there a fix? Remi ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recording calls after queues?
Greetings, List. With my current setup, I record all incoming calls to my queues. My problem is that once a call is transferred out of a queue, recording stops. How can I make it so recording continues even after a call is transferred? If you need me to post any dialplan or conf logic, please ask. Thanks, Jay ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blip every 30 seconds?
AHHA! The PRI was not plugged in (system still in testing) so the timing was off. As soon as we plugged the PRI cable in the blips went away.. On 8/2/07, Matt [EMAIL PROTECTED] wrote: Strange issue when I record a file from a phone to the asterisk system I get a blip in the recording every 30 seconds. It's a very small blip, but it is there.It seems like it's only if I'm recording, not when I'm playing back that the issue happens. My SATA drives, ETH0, and my Sangoma card are all on seperate IRQs. Any thoughts on what might be causing this and how to stop it? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blip every 30 seconds?
Joe acquisto wrote: Telephone conversations that are being recorded, are supposed to beep periodically, to alert/remind the recorded person that the conversation is being recorded. Perhaps that is what you are hearing? joe a. On 8/2/2007 at 8:47 AM, Matt [EMAIL PROTECTED] wrote: Strange issue when I record a file from a phone to the asterisk system I get a blip in the recording every 30 seconds. It's a very small blip, but it is there.It seems like it's only if I'm recording, not when I'm playing back that the issue happens. My SATA drives, ETH0, and my Sangoma card are all on seperate IRQs. Any thoughts on what might be causing this and how to stop it? What version of asterisk are you using? Senad ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] radius support
yonoko molomo wrote: hi, how to add radius support to asterisk 1.4.5? i do make menuselect and i do not see any module or option related to radius, pam, authenticacion or similar. any ideas? thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://www-rocq.inria.fr/who/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] uptime script?
Can someone point me to an agi script that will read back the asterisk uptime, if such a thing exists? - Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple pbxes, multiple domains, same user ids?
You may want to consider the multi-tenant version of Thirdlane's PBX Manager (www.thirdlane.com). I've been using for a long time and very happy with both single and multi-tenant versions. Benjamin Jacob wrote: Anthony Francis wrote: Hello good ppl, A couple of questions for multiple pbxes 1. Is it possible to support multiple pbxes in one Asterisk box(using contexts, etc.)? 2. Can we use the "domain" field in sip.conf to specify the different domains for sip users, having one domain for each pbx? I just tried registering two xlites, with different domain names (with the same specified in sip.conf). But, Asterisk maintains the registration of the latest registree!! thats really sad for me . Any work around for this one(multiple pbx)? I would be zapped and amazed if multiple pbx isn't possible in Asterisk. Help anyone? cheers - Ben. you have to do different sip-ids, I am guessing you are probably using the extension #, you dont need to do that. What do you mean by multiple-pbx's anyway? I hope you don't mean multiple instances of *.What I am sure you mean is multiple dial plans, and yes, * is multi-tenant friendly. What we do for uniqueness is use the last 8 digits of the device mac addr or other unique number followed by a dash "-" followed by the extension number. Anthony Thanks Anthony. I definitely don't mean multiple instances of asterisk. Multiple dial plans, hmm.. yes.. in a way. Multiple pbx ... in short, provide pbxes for two entirely different organizations, say, Microsoft and IBM (can i use these names in here? ;-) ). Each would have many extensions, but each office can have identical extensions, e.g. you can have extensions 4001 in both. But one would be [EMAIL PROTECTED] and the other would be [EMAIL PROTECTED] . [EMAIL PROTECTED] should be able to call any user within Microsoft. To step outside the organization, you would put in some logic(dialplans). So, i want to have pbx for microsoft and another pbx for IBM. Is it possible to have two or more pbxes within one Asterisk instance. Hope you got my point. cheerz - Ben. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blip every 30 seconds?
Gordon Henderson wrote: On Thu, 2 Aug 2007, Joe acquisto wrote: Telephone conversations that are being recorded, are supposed to beep periodically, to alert/remind the recorded person that the conversation is being recorded. You really ought to qualify this with the country and the relevant laws that you think require it ... In the UK there is no such law/rule/supposition. You don't even have to tell the other party the call is being recorded - just one person in the conversation needs to know. (although your calls may be recorded for training purposes ... whatever they are!) So if you call me, then your call may be recorded. Or it may not be. You'll never know ... Gordon Joe never mentioned law, he just said supposed to. I believe that this is common courtesy really, so you should or are supposed to probably make sense. At least to my way of thinking. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware that can ring my phone?
Wow! Thank you so much, James - you have certainly clarified lots of things in my mind. You are correct about me overlooking the feedback issue (with the el-cheapo device). I see that I have to learn. This world of VoIP is new and mind boggling - to me. Thanks, Lynn --- James FitzGibbon [EMAIL PROTECTED] wrote: On 8/1/07, Linux Lover [EMAIL PROTECTED] wrote: This SOHO PBX box won't interop with Asterisk because it doesn't speak any of the protocols that Asterisk does. This box I tend agree with your evaluation. Still, I was thinking that since all these el-cheapo SOHO PBX boxes support manual attendant call transfer, what's to prevent Asterisk from mimicking an attendant by sending proper DTMF signals and make this box transfer the call to the single analog phone in the business? That is, Asterisk will connect (via RJ-11) to the unit as the attendant's phone, and my real phone (only one in the system) will connect via a second RJ-11 (there could be 4 of them). Or is Asterisk not capable of sending DTMF signals over an RJ-11 connection? You can send arbitrary DTMF over any of Asterisk's channels from the dialplan. I just figured that this level of integration was a bit deeper than you were looking for as a first project. It would be an interesting experiment, to be sure. The biggest issue I'd think would be feedback - you can send the DTMF along the wire, but how do you know that the SOHO box interpreted it correctly? If the only feedback is designed for a human (i.e. auditory), then interpreting those cues with Asterisk would be non-trivial. Do I undestand correctly that with this solution, I will still be able to connect to my analog Verizon phone line with the SIP phone? That is, the outside world will see my phone as an ordinary phone, when in fact I am using a SIP phone? If so, that means that Asterisk does all the magic behind the scene, right? Yes, your Verizon POTS line would go into a FXO port in your server (which in Asterisk would be referenced as the channel Zap/1 - zaptel being Asterisk's TDM driver) and your SIP phone would connect via your standard office network and be referenced as SIP/whateverusernameyouwant. A very simplistic example of bridging a call would be: [from-verizon] exten = s,1,Dial(SIP/whateverusername) Assuming that you'd configured zaptel to route calls that come in on the FXO port to the Asterisk context named from-verizon, then any such calls would immediately cause Asterisk to ring your SIP phone, and if answered to bridge the two calls together. A more complex example that makes them press one to call you and otherwise lets them leave a message: [from-verizon] exten = s,1,Background(Press1ToTalkOr2ToLeaveAMessage) exten = s,n,WaitExten(10) ; timeout exten = t,1,Goto(vm,1) ; invalid exten = i,1,Goto(vm,1) ; press 1 exten = 1,1,Dial(SIP/101,20) exten = 1,n,Goto(vm,1) ; press 2 exten = 2,1,Goto(vm,1) ; all voicemail activity ends up here exten = vm,1,VoiceMail(u101) exten = vm,n,Hangup [from-officephone] exten = *98,1,VoiceMailMain extne = *98,n,Hangup Assuming you've now set up your SIP phone as extension 101, this would play a sound file saying press 1 to talk to 2 to leave a message. If they press 1, your SIP phone rings. If they press 2, they go to voicemail. If they wait 10 seconds without pressing anything, or press something other than 1 or 2, they also go to voicemail. If they press 1 to dial your phone and you don't pick up after 20 seconds, they go to voicemail. On your deskphone (could just as easily be a SIP softphone if you prefer), you can dial *98 to log in and pick up your new voicemail messages. Hope that demystifies some of what you're trying to do. -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use stun server?
hi again.well i have been trying to know what is the relationship between asterisk and stun. what i mean is, i understand that a client requests stun server to know whether its behind a nat or not. if its not, then its ok. if it is behind nat, then what? Now client knows what kind of nat it is behind, what is the roll of asterisk in it. asterisk already knows client's public ip whether its behind nat or not, if the client is registered. So how does stun simplify things if there are nat problems. After requesting stun server and recieving the required information from stun server.what happens next? I hope im clear in stating my problem. Hope to hear from you soon On 8/1/07, SIP [EMAIL PROTECTED] wrote: No... there's no STUN server built into Asterisk. Asterisk handles NAT in a different way... and is an endpoint rather than a proxy, so it doesn't really NEED STUN built into it. However, we run a STUN server on the same machine as an Asterisk server and see nothing in terms of load increase. STUN's footprint is rather negligible. N. Rizwan Hisham wrote: Ok thanx. One more thing to ask is: does asterisk has a stun server implemented in it or not. i mean does asterisk contain a stun server and provides any application which can be used for enabling the stun server in asterisk? On 8/1/07, *SIP* [EMAIL PROTECTED] mailto: [EMAIL PROTECTED] wrote: STUN is a pretty simplistic server. There's nothing else that needs to be configured on the STUN server side. It's pretty much either running or it's not. Just start plugging in the server to your clients and give it a whirl. It should work. N. Rizwan Hisham wrote: Hi all, This is the first time i am using stun with asterisk for nat problems. I have read the rfc which describes how stun works. i didnt have any problems understanding it. I have also intalled the stun server called stund which i downloaded from sourceforge. I have seen on the list that most people use stund here. I have started the stun server and its running silently. Now i dont know what to do next. Is there something more to it, like configuration files which i can use for special configuration for asterisk, or is there not. How do i proceed, if there is nothing more to configure in stun, does that mean i can start configuring my clinets (xten and sipura) to use stun server? -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com http://www.axvoice.com http://www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com http://www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blip every 30 seconds?
On Thu, 2 Aug 2007, Joe acquisto wrote: Telephone conversations that are being recorded, are supposed to beep periodically, to alert/remind the recorded person that the conversation is being recorded. You really ought to qualify this with the country and the relevant laws that you think require it ... In the UK there is no such law/rule/supposition. You don't even have to tell the other party the call is being recorded - just one person in the conversation needs to know. (although your calls may be recorded for training purposes ... whatever they are!) So if you call me, then your call may be recorded. Or it may not be. You'll never know ... Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Receiving SIP calls without registeration and dynamic IP address
Hi List; How can I configure asterisk to receive a call from SIP end point without being registered at asterisk and its IP address is dynamic, and authentication to be based on the username and password or any other string? I know that if I place the host with static IP then no need to register, but what if the voip gateway was having dynamic IP and I do not need to register on asterisk, but I need to send the call directly and asterisk authenticate the device based on the username and password or any possible string? Regards, --- ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460 Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] H.323
Hi List; Did any one tried the H.323 module? How much it is stable and work fine? Regards, ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460 Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. http://tv.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Receiving SIP calls without registeration and dynamic IP address
Bilal, The purpose of registration is to establish a contactability/reachability URI information in the registrar dynamically. If you have a static IP on both ends you can nail up an IP-trusted peer session / SIP trunk. If not, some form of registration will be required. Registration does not necessarily require a username and password; in fact, it is rarely sent with the registration anyway. Instead it is usually sent as a response to a 407 proxy challenge in subsequent requests, unless the REGISTER message is interrogated with that prior to being accepted, which depends on how you have the UAS configured. Other than that, not quite sure what you're asking precisely... -- Alex On Thu, 2 Aug 2007, bilal ghayyad wrote: Hi List; How can I configure asterisk to receive a call from SIP end point without being registered at asterisk and its IP address is dynamic, and authentication to be based on the username and password or any other string? I know that if I place the host with static IP then no need to register, but what if the voip gateway was having dynamic IP and I do not need to register on asterisk, but I need to send the call directly and asterisk authenticate the device based on the username and password or any possible string? Regards, --- ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460 Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] radius support
Hi, http://www-rocq.inria.fr/who/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html Thanks, I have already seen that document before but it did not help much to have a better understanding to set up radius with asterisk. In 4.3 it is written: Asterisk has been patched along with the previously decribed PAM radius module. But I was not able to find that patch. In any case, is pam_radius not supported by asterisk without patching it? I thought it was supported (i am using stable version 1.4.5 of asterisk) In the next sentence, A discussion on how to provide RADIUS functions to Asterisk can be found here, along with the patch : http://bugs.digium.com/view.php?id=5424; there is that link, but again, it is not helping me to understand how to do this. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk1.2 to 1.4 g711a fax
hi, i have problem with pass-through faxing with this scenario hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.2.X(xen virtual) - linksys ATA i can fax to fax2mail on hylafax but after upgrade asterisk2 to 1.4 faxing is not working hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.4.X(xen virtual) - linksys ATA configuration is same do you hava any idea what is changed in 1.4 in g711 pass-through faxing? thanks marek ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use stun server?
Honestly, it's really up to the client how it handles information from STUN. Ideally, what will happen is that it will modify its Contact headers and SDP information to include the STUN-discovered IP address and port. In so doing, when it sends out a request to another server, that server will then know the proper IP address to use to send data back to the UA. This is primarily of importance when you are using SER/OpenSER as a SIP proxy, or have Asterisk set to canreinvite=yes What happens is that this allows clients to directly talk to each other using publicly-addressable IP addresses, taking Asterisk out of the equation except for passing signaling information. It can save bandwidth. It can ease Asterisk load. Etc, etc. If you have canreinvite=no set on your Asterisk server, and you're using Asterisk for your SIP communications, then STUN will still inform the UA to rewrite its appropriate headers, but you'll see no real difference. Audio will still be bridged by the Asterisk box. Your bandwidth won't change. Etc, etc. It all really depends on what you want to get out of this whole thing and what your overall network design is. N. Rizwan Hisham wrote: hi again.well i have been trying to know what is the relationship between asterisk and stun. what i mean is, i understand that a client requests stun server to know whether its behind a nat or not. if its not, then its ok. if it is behind nat, then what? Now client knows what kind of nat it is behind, what is the roll of asterisk in it. asterisk already knows client's public ip whether its behind nat or not, if the client is registered. So how does stun simplify things if there are nat problems. After requesting stun server and recieving the required information from stun server.what happens next? I hope im clear in stating my problem. Hope to hear from you soon On 8/1/07, *SIP* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: No... there's no STUN server built into Asterisk. Asterisk handles NAT in a different way... and is an endpoint rather than a proxy, so it doesn't really NEED STUN built into it. However, we run a STUN server on the same machine as an Asterisk server and see nothing in terms of load increase. STUN's footprint is rather negligible. N. Rizwan Hisham wrote: Ok thanx. One more thing to ask is: does asterisk has a stun server implemented in it or not. i mean does asterisk contain a stun server and provides any application which can be used for enabling the stun server in asterisk? On 8/1/07, *SIP* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: STUN is a pretty simplistic server. There's nothing else that needs to be configured on the STUN server side. It's pretty much either running or it's not. Just start plugging in the server to your clients and give it a whirl. It should work. N. Rizwan Hisham wrote: Hi all, This is the first time i am using stun with asterisk for nat problems. I have read the rfc which describes how stun works. i didnt have any problems understanding it. I have also intalled the stun server called stund which i downloaded from sourceforge. I have seen on the list that most people use stund here. I have started the stun server and its running silently. Now i dont know what to do next. Is there something more to it, like configuration files which i can use for special configuration for asterisk, or is there not. How do i proceed, if there is nothing more to configure in stun, does that mean i can start configuring my clinets (xten and sipura) to use stun server? -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com http://www.axvoice.com http://www.axvoice.com http://www.axvoice.com http://www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] Receiving SIP calls without registeration and dynamic IP address
On Thu, 2007-08-02 at 08:11 -0700, bilal ghayyad wrote: How can I configure asterisk to receive a call from SIP end point without being registered at asterisk and its IP address is dynamic, and authentication to be based on the username and password or any other string? I think you're confused here... registration has nothing to do with a SIP device being able to send calls to Asterisk. A SIP devices registers with Asterisk so that Asterisk knows where to send calls going *to* the device. For calls coming into Asterisk, the SIP channel driver first looks at all the users in sip.conf (you know, everything set with type=user or type=friend). It matches on the name in square brackets as the SIP username, and the password on the secret= line. If the device authenticates correctly, the call gets sent to the dialplan in the context specified by the context= line. As an example, let's say we had the following in sip.conf: [test] type=user secret=abc123 context=hamburger If any SIP device were to come along and authenticate with the username test and the password abc123, Asterisk would accept the call and send the call to the [hamburger] context in the dialplan. Asterisk would do this *whether or not* the device had registered. Now, as I understand it, if Asterisk can't find any users (or friends) that match, it then goes looking through the list of peers, trying to match the host= field to the IP address of the device that's sending the call to Asterisk. Hopefully that clarifies things for you. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use stun server?
On Thu, 2 Aug 2007, Rizwan Hisham wrote: hi again.well i have been trying to know what is the relationship between asterisk and stun. what i mean is, i understand that a client requests stun server to know whether its behind a nat or not. if its not, then its ok. if it is behind nat, then what? Now client knows what kind of nat it is behind, what is the roll of asterisk in it. asterisk already knows client's public ip whether its behind nat or not, if the client is registered. So how does stun simplify things if there are nat problems. There is no relationship between asterisk and STUN. After requesting stun server and recieving the required information from stun server.what happens next? I hope im clear in stating my problem. I'm not a STUN/SIP protocol gury by any means, but this is my understanding (and it might be a bit simplistic) When something communicates with something else using SIP, the sending device (eg phone) puts it's own IP address inside the SIP data packet. That IP address is the IP address of the device - it doesn't know anything about anything else, just the IP address it has. This would work well if NAT hadn't been invented, unfortunately it was. The listening side (eg. asterisk), extracts this IP address and uses it to send data back. So if the originating device is behind NAT, and it's on (eg) 192.168.0.42 then the other end, gets that IP address and tries to send data back to it. Which, as 192.168.0.42 is on a private network, it can't do. Oops. So the original device uses a STUN server to poke a few bytes over the interweb and the STUN server replys back with some information - such as the real external IP address and port numbers it's using. The STUN server is a tiny bit of software running on a host somewhere with a real IP address (or 2!) and is (or can be) quite independant of the asterisk server. Original device can then put those values returned from the STUN server inside the SIP data packets (rather than it's 'real' natted IP address) and send them off to the other end, which can then use them to send the replys back to. The device should only need to access the STUN server once in it's life, but devices periodically check, just in-case things have changed. They do not relay data through the STUN server. So that's for device to asterisk box. Asterisk boxes are supposed to be directly connected to the Internet with no NAT and a real live IP address. (or at least that's the best possible way to do it!) If they aren't ... Then the first thing you need to do is arrange port-forwarding from the firewall to the asterisk box. You'll need to forward the ports you need - eg. for SIP it might be 5060-5069 and for RTP it might be 1-2. But the asterisk server still needs to know what it's real external IP address is so it can put that in the SIP packets rather than it's own NATted address, and as asterisk can't use a STUN server, you need to explicitly tell it - this is in the sip.conf file and looks like: nat=yes localnet=192.168.2.0/24 externip=1.2.3.4 So now the asterisk server knows that anything that originates from the local network doesn't need to be translated, but anything going out needs to have the SIP data re-written with the real external IP address. Now (AIUI) some SIP proxys can look inside SIP data packets and see that the IP address given by the device is not the same as the IP address that the packet came from and adjust things accordingly.. Asterisk, not being a SIP proxy doesn't do this, so if your phone is talking to a server via a proxy, then you may not need to tell the phone about a STUN server. The people running the asterisk+SIP proxy will tell you if this is the case. I'm sure there was a perfectly good reason for encoding the devices IP address inside the SIP data when they invented it, but right now, I can't think why... See http://www.ietf.org/rfc/rfc3261.txt for the details! Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blip every 30 seconds?
On Thu, 2 Aug 2007, Steve Totaro wrote: Gordon Henderson wrote: On Thu, 2 Aug 2007, Joe acquisto wrote: Telephone conversations that are being recorded, are supposed to beep periodically, to alert/remind the recorded person that the conversation is being recorded. You really ought to qualify this with the country and the relevant laws that you think require it ... In the UK there is no such law/rule/supposition. You don't even have to tell the other party the call is being recorded - just one person in the conversation needs to know. (although your calls may be recorded for training purposes ... whatever they are!) So if you call me, then your call may be recorded. Or it may not be. You'll never know ... Gordon Joe never mentioned law, he just said supposed to. I believe that this is common courtesy really, so you should or are supposed to probably make sense. At least to my way of thinking. You're right. But there is not even a hint of supposition in the UK. You're not supposed to do anything, and from what I gather, in some orginisations supposing a common coutresy would appear to be discouraged. At best you might get a message on their web site, and very occasionally a recording (which, given that you're paying for the call adds to the irritation), that your calls may be recorded for training purposes, but more often than not, you don't get told (as there is no legal obligation in the UK!) yet you know that every time you call any major entity - bank, insurance co, mortgage, finance, telco, ISP, etc. the call will be recorded. Gordon (In our brave new recorded and monitored, big-brother world) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pri call by call trunking?
We spent a considerable amount of time getting an A101 up and running. Try to find out what type of switch you are connecting to. In our case, we were working against a Nortel. For some reason, if we used ni2, it would not work. Finally setting the switchtype to 5ess or DMS100 would work and now everything sings. Hope that helps. Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Wednesday, August 01, 2007 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] pri call by call trunking? Call Sangoma On 8/1/07, Erik Anderson [EMAIL PROTECTED] wrote: On 8/1/07, John covici [EMAIL PROTECTED] wrote: I had some troubles -- try setting the timing parameter to 0 (second one in your span) and see if that helps. If I'm reading the docs correctly, this param should only be set to 0 if you *never* want to use the T1 connected to this port for timing. That's not the case in my situation, as I need to be syncing with the telco's clock. That said, in the interest of troubleshooting, I did try setting it to zero - this didn't fix the problem. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI SAY TIME
Hello all, Can anyone help me with SAY TIME. Every time I ask to say time, it gives me wrong time. I want the system to say time, what ever I give to say. Is it possible? Cheers, Nitesh ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H.323
Hi there, I have use the H.323 module that comes with asterisk-addons and i consider it (so far) VERY stable for my needs. Im talking about 10,000 minutes at month , + or - , and never had a crash or something bad about it. Personally, i recommend it, -- J. P. rakh at slackware-es dot com bilal ghayyad wrote: Hi List; Did any one tried the H.323 module? How much it is stable and work fine? Regards, ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460 Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. http://tv.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Teliax Quality of Service
Asterisk Users, I recently ran into some problems with the quality of service with Teliax. This occurred on August 1, 2007 with a dropped outbound call, audio quality isse on the callee side- not hearing me well on callee side, and sending DTMF tones (configured for RFC2833). Am I the only Teliax customer having this problem? It seems like when I am ready to go live with my Asterisk PBX System, I run into quality of service issues with the SIP provider. Who should I go with that would guarantee me quality service just like an analog line? _ See what youre getting into before you go there http://newlivehotmail.com/?ocid=TXT_TAGHM_migration_HM_viral_preview_0507 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax Quality of Service
There is a strong possibility that the problem is on your side. Are you using a cable or dsl? What are your download and upload speeds? Are you doing any kind of traffic shaping? You will not get a guarantee of QoS from any provider. They cannot control what is happening on your end or what happens on the public internet. Maybe if you put in a point to point to the provider, then they might consider it. If you are seriously considering doing business using VoIP, then I would reconsider unless your internet provider is providing the VoIP service and they observe QoS on their equipment. Otherwise, you can never be sure what the quality will be at any given time. Weigh the saving against the cost of dropped or garbled calls. Thanks, Steve John Meksavan wrote: Asterisk Users, I recently ran into some problems with the quality of service with Teliax. This occurred on August 1, 2007 with a dropped outbound call, audio quality isse on the callee side- not hearing me well on callee side, and sending DTMF tones (configured for RFC2833). Am I the only Teliax customer having this problem? It seems like when I am ready to go live with my Asterisk PBX System, I run into quality of service issues with the SIP provider. Who should I go with that would guarantee me quality service just like an analog line? _ See what you’re getting into…before you go there http://newlivehotmail.com/?ocid=TXT_TAGHM_migration_HM_viral_preview_0507 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax Quality of Service
John Meksavan wrote: Asterisk Users, I recently ran into some problems with the quality of service with Teliax. This occurred on August 1, 2007 with a dropped outbound call, audio quality isse on the callee side- not hearing me well on callee side, and sending DTMF tones (configured for RFC2833). Am I the only Teliax customer having this problem? It seems like when I am ready to go live with my Asterisk PBX System, I run into quality of service issues with the SIP provider. Who should I go with that would guarantee me quality service just like an analog line? If you want service to be as reliable as the PSTN then you have to use the PSTN. I feel that sending calls over the Internet is just silly if you want as close to %100 uptime as you can. My customers use PRIs with VoIPoInternet as a failover in case the PRI goes down or all channels are in use on the PRI. I am not saying that VoIP is unreliable. It is very reliable -- when you control the lines and routers between you and the PSTN. I'm saying that the Internet is not reliable. My customers route calls over point to point T-1s all the time with no issues. Teliax seems to one of the better ITSPs. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax Quality of Service
At 09:23 AM 8/2/2007, you wrote: I recently ran into some problems with the quality of service with Teliax. This occurred on August 1, 2007 with a dropped outbound call, audio quality isse on the callee side- not hearing me well on callee side, and sending DTMF tones (configured for RFC2833). Am I the only Teliax customer having this problem? Teliax has been quite good. I was having problems the last 2 days and they confirmed that they are working on fixing something. I've been using IP for all my outgoing calls for the last couple of years and other than being ripped off by a couple of vendors and the occasional connection problem it's saved me large amounts of money, more than what I lost when the 2 providers refused to return my deposits and then went under, but I do have ways to get dial tone on my POTS lines for those times when it all goes to heck. Ira ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Royalty for On Hold Music ?
On Tue, 31 Jul 2007, Steve Kennedy wrote: What if the radio is on in the background when I make a call ? is that rebroadcasting ? kind of gets blurry on the definitions there. That's not as you're listening to it and not trying to rebroadcast. I've not been following this thread closely, so apologies if this has already been covered. I had a summer job many years ago (early '90s) for the organisation responsible for collecting royalties in Ireland (IMRO). My recollection is probably a bit off, but the situation was that: - if you played copyrighted music on your phone system you needed a license which was scaled on the number of external channels on your phone system - if you had copyrighted music playing in the background in your office/shop/workplace then you needed a license which was scaled on the number of people working in your office/shop/workplace The reasoning behind both was that the employer was making (or allowing) the music available to third parties which was classed as a performance in a public place, which incurs a royalty fee (public == anything that's not domestic). It didn't matter whether the music came from TV, radio or a recording (and royalties were also levied on the TV, radio and recording companies). IIRC The licenses were typically an annual fee on the order of (back then) about IEP 100-200 (now EUR 127-254). AFAIR the situation was similar in the UK, where the Performing Rights Organisation (PRO) were the equivalent body. -Ronan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE220B
Has anyone ever had any problem with the TE220B card with it showing up as four ports instead of two. I RMA'd the first one with the retailer (Digium tech advice), but I just got another brand new card and it is coming up as four ports again. The card identifier is showing 0420 when I do lspci. Has this happened to anyone and if so is there a fix? I don't know why, but your PCI subvendor ID seems to be set to the wrong value. Unfortunately, it's probably not modifiable with the tools at your disposal. There is a way to make this card work properly, but it will make any 420's in the system act like 220's as well. Open up wct4xxp/base.c and search for 0220. You should find yourself in the pci_device_id structure. You'll need to modify the 0220 to be 0420 and you'll also need to comment out the line above it that contains 0420. After that, run make install and reload the wct4xxp.ko driver. Your TE220 should work fine with those modifications. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax Quality of Service
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ira Sent: Thursday, August 02, 2007 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Teliax Quality of Service At 09:23 AM 8/2/2007, you wrote: I recently ran into some problems with the quality of service with Teliax. This occurred on August 1, 2007 with a dropped outbound call, audio quality isse on the callee side- not hearing me well on callee side, and sending DTMF tones (configured for RFC2833). Am I the only Teliax customer having this problem? Teliax has been quite good. I was having problems the last 2 days and they confirmed that they are working on fixing something. I've been using IP for all my outgoing calls for the last couple of years and other than being ripped off by a couple of vendors and the occasional connection problem it's saved me large amounts of money, more than what I lost when the 2 providers refused to return my deposits and then went under, but I do have ways to get dial tone on my POTS lines for those times when it all goes to heck. I confused by this. Don't ITSP's have redundancy? Don't they have multiple edge systems for accepting incoming calls? Don't their multiple edge systems have multiple interfaces, connected to multiple subnets, via multiple switches? And, don't they have multiple upstream providers? About the only thing that could go wrong that affects all service like this would be a badly pushed out software update, affecting all systems? Doug. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Receiving SIP calls without registeration and dynamic IP address
Hi Alex; Kindly find my answers below preceeded by ( * ). Bilal, The purpose of registration is to establish a contactability/reachability URI information in the registrar dynamically. * What is the URI? If you have a static IP on both ends you can nail up an IP-trusted peer session / SIP trunk. If not, some form of registration will be required. Registration does not necessarily require a username and password; in fact, it is rarely sent with the registration anyway. Instead it is usually sent as a response to a 407 proxy challenge in subsequent requests, * Who send the 407 proxy challenge and what is that 407 proxy challenge? unless the REGISTER message is interrogated with that prior to being accepted, which depends on how you have the UAS configured. * What is the UAS, also I did not get u in this paragraph. Other than that, not quite sure what you're asking precisely... * I was asking if the endpoint send a call, and it has a username and password typical to that configured in SIP.conf file, then should this end point being registered or not? Regards Bilal Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=listsid=396545433 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple pbxes, multiple domains, same user ids?
Forums wrote: You may want to consider the multi-tenant version of Thirdlane's PBX Manager (www.thirdlane.com). I've been using for a long time and very happy with both single and multi-tenant versions. Benjamin Jacob wrote: Anthony Francis wrote: Hello good ppl, A couple of questions for multiple pbxes 1. Is it possible to support multiple pbxes in one Asterisk box(using contexts, etc.)? 2. Can we use the domain field in sip.conf to specify the different domains for sip users, having one domain for each pbx? I just tried registering two xlites, with different domain names (with the same specified in sip.conf). But, Asterisk maintains the registration of the latest registree!! thats really sad for me . Any work around for this one(multiple pbx)? I would be zapped and amazed if multiple pbx isn't possible in Asterisk. Help anyone? cheers - Ben. you have to do different sip-ids, I am guessing you are probably using the extension #, you dont need to do that. What do you mean by multiple-pbx's anyway? I hope you don't mean multiple instances of *.What I am sure you mean is multiple dial plans, and yes, * is multi-tenant friendly. What we do for uniqueness is use the last 8 digits of the device mac addr or other unique number followed by a dash - followed by the extension number. Anthony Thanks Anthony. I definitely don't mean multiple instances of asterisk. Multiple dial plans, hmm.. yes.. in a way. Multiple pbx ... in short, provide pbxes for two entirely different organizations, say, Microsoft and IBM (can i use these names in here? ;-) ). Each would have many extensions, but each office can have identical extensions, e.g. you can have extensions 4001 in both. But one would be [EMAIL PROTECTED] and the other would be [EMAIL PROTECTED] . [EMAIL PROTECTED] should be able to call any user within Microsoft. To step outside the organization, you would put in some logic(dialplans). So, i want to have pbx for microsoft and another pbx for IBM. Is it possible to have two or more pbxes within one Asterisk instance. Hope you got my point. cheerz - Ben. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Right, but in asterisk it is not done this way, you just use dial-plan contexts to separate the entities, I have over 200 unique companies dial-plans spread over 12 asterisk boxes and every single one of them has a base set of extensions that are exactly the same. What you do is have a master context for incoming calls that matches any full dids for the companies, when matched, the call is transferred into that customers context. Then in the sip.conf you make sure that the context= is set to that customers context, and you are good to go. Anthony ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dtmf get data
Greetings, We have a handlewelcome.agi script which handles every new caller. For every new call we play a welcome message and ask the caller to enter a four digit code .. something on the lines Welcome... please enter the four digit number Our asterisk java agi script calls a function getData() with parameters such as the gsm file to play the message and the number of dtmf characters to receive. (the getData() call maps to the asterisk cmd get data) We have noticed that if a user keyed in the four digit code while the message is being played. the dtmf char received is only one. i.e if i key in 1007 before the message is fully played we get only 1 char. but if i wait for the whole message to complete then there is no problem we receive the complete code 1007 . But i remember that before we could enter and receive the dtmf digits correctly even if the message was playing. Best Regards, Shivram ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Receiving SIP calls without registeration and dynamic IP address
On Thu, 2 Aug 2007, William Moore wrote: If you are only *SENDING* calls to asterisk and not receiving, you do not need to send a registration. You only need to send a registration if you want to *RECEIVE* calls from asterisk. That's not necessarily true. Asterisk isn't going to just let any old IP address anywhere send a call through it; it depends on how you configure the relevant SIP peers. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Receiving SIP calls without registeration and dynamic IP address
On Thu, 2 Aug 2007, Alex Balashov wrote: On Thu, 2 Aug 2007, William Moore wrote: If you are only *SENDING* calls to asterisk and not receiving, you do not need to send a registration. You only need to send a registration if you want to *RECEIVE* calls from asterisk. That's not necessarily true. Asterisk isn't going to just let any old IP address anywhere send a call through it; it depends on how you configure the relevant SIP peers. Although, Asterisk will probably behave as most UACs do here and simply challenge the INVITE, rather than demand a REGISTER, so on second thought you are correct. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Receiving SIP calls without registeration and dynamic IP address
* I was asking if the endpoint send a call, and it has a username and password typical to that configured in SIP.conf file, then should this end point being registered or not? If you are only *SENDING* calls to asterisk and not receiving, you do not need to send a registration. You only need to send a registration if you want to *RECEIVE* calls from asterisk. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dtmf get data
Shivaram U wrote: Greetings, We have a handlewelcome.agi script which handles every new caller. For every new call we play a welcome message and ask the caller to enter a four digit code .. something on the lines Welcome... please enter the four digit number Our asterisk java agi script calls a function getData() with parameters such as the gsm file to play the message and the number of dtmf characters to receive. (the getData() call maps to the asterisk cmd get data) We have noticed that if a user keyed in the four digit code while the message is being played. the dtmf char received is only one. i.e if i key in 1007 before the message is fully played we get only 1 char. but if i wait for the whole message to complete then there is no problem we receive the complete code 1007 . But i remember that before we could enter and receive the dtmf digits correctly even if the message was playing. Best Regards, Shivram Before what? An upgrade? A change in code? The answer to that question may reveal a bug or just an answer to your question. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MySQL + Realtime + SIP Registration
I have read and followed as much as I can find but I am missing something. What I want to do is get as much as I can running from mysql and keep the *.conf files for static things. So I have setup a SIP users/peers table in a mysql database and I have populated it with a few peers. I have configured asterisk addons and from the asterisk CLI I am able to search the sip users / peers tables using the realtime load command. This is after i added sipusers = mysql,asterisk,sip_users to my extconfig.conf file. However I don't know what to do to get asterisk to look at that table when a request to register comes from a sip peer. I understand that sipusers and sip peer are contradictory but they are all defined as peers. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI/T1 data rate...
Hi all, First, this is not my first PRI/T1 Asterisk deployement. Did several with Bell, Telus, AllStream, Rogers but this is my first with Videotron. Just spoke with the person taking the order and on top of the standard settings (switch, coding,...) she asked me about data rate (56k or 64k). Since I have never been asked this question before and can find anything relevant in the configuration files, I would like to have your inputs on this. Thanks, Andre Courchesne ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI/T1 data rate...
B-chans should be 64k. That is a strange question indeed. Thanks, Steve Totaro Andre Courchesne - Consultant wrote: Hi all, First, this is not my first PRI/T1 Asterisk deployement. Did several with Bell, Telus, AllStream, Rogers but this is my first with Videotron. Just spoke with the person taking the order and on top of the standard settings (switch, coding,...) she asked me about data rate (56k or 64k). Since I have never been asked this question before and can find anything relevant in the configuration files, I would like to have your inputs on this. Thanks, Andre Courchesne ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording calls after queues?
On 8/2/07, Jay Moore [EMAIL PROTECTED] wrote: With my current setup, I record all incoming calls to my queues. My problem is that once a call is transferred out of a queue, recording stops. How can I make it so recording continues even after a call is transferred? If you need me to post any dialplan or conf logic, please ask. Explicity invoke MixMontitor() in your dialplan before calling Queue() instead of using monitor-format=whatever in queues.conf. If you get to some point where you want to stop the recording, call StopMixMonitor(). -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] callback and bridge problem
Greetings, i've been posted a message to this list in july, which had one response. Thanks for that idea! Unfortunately asterisk is only a hobby, and did not have much time dealing with the problem since. My original letter was long, i wouldn't post it again, the archive url is http://archives.free.net.ph/message/20070710.053008.c02209c0.en.html Since than i've upgraded to 1.4.8 from 1.2 series, i thought this might help. It did not. Answering to the question asked from me in july, no, i'm not behind nat, and i did not have reinvite=yes in my configs. I've put it into the sip.conf, tried, but the call hung up again. I'd be greatful for more ideas of solving the problem. Fresh logs when hanging up, from asterisk console: -- SIP/neophonex99-out-08213ac8 is making progress passing it to SIP/neophonex57-out-081e8a78 [Aug 2 21:54:51] WARNING[24739]: chan_sip.c:11948 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '[EMAIL PROTECTED]'. Giving up. -- SIP/neophonex99-out-08213ac8 answered SIP/neophonex57-out-081e8a78 -- Native bridging SIP/neophonex57-out-081e8a78 and SIP/neophonex99-out-08213ac8 [Aug 2 21:54:57] WARNING[24739]: chan_sip.c:11948 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '[EMAIL PROTECTED]'. Giving up. == Spawn extension (internal, 9520620*, 3) exited non-zero on 'SIP/neophonex57-out-081e8a78' [Aug 2 21:54:57] NOTICE[24749]: pbx_spool.c:351 attempt_thread: Call completed to SIP/[EMAIL PROTECTED] Thanks for any help Adam ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI/T1 data rate...
Steve Totaro wrote: B-chans should be 64k. That is a strange question indeed. For PRI, agreed. This is, however, a common question when provisioning channelized T1 services, since the B channels on robbed-bit T1's are really only 56K since the lowest bit is robbed for signalling. -forrest ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI/T1 data rate...
Forrest W Christian wrote: Steve Totaro wrote: B-chans should be 64k. That is a strange question indeed. For PRI, agreed. This is, however, a common question when provisioning channelized T1 services, since the B channels on robbed-bit T1's are really only 56K since the lowest bit is robbed for signalling. -forrest I knew someone would have an explanation that makes sense. I have NEVER done anything but PRI from the Telco. Wouldn't the question of signaling and switchtype negate the need to ask for data rate? Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unicall and Private CID
It seems the problem with Unicall and Nextel is also present in Asterisk 1.2 and not only in 1.4. I decided to downgrade from 1.4.9 to 1.2.23 so the customer could have CID and calls from Nextel but today he told me that they cannot receive any calls from Nextel, they get a busy tone every time. I downloaded the following from softswitch: http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.3.tgz http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre11/libmfcr2-0.0.3.tar.gz http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre11/libsupertone-0.0.2.tar.gz http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre11/libunicall-0.0.3.tar.gz http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre11/asterisk-1.2.x/chan_unicall.c http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre11/asterisk-1.2.x/channels_Makefile.patch The patch file fails in three places but I patched by hand. All other calls come in and out, only calls from Private CID (like Nextel) get a busy tone all the time. Could it be that this is something that got broken on more recent versions of libmfcr2? I have other systems installed over two years ago that do not have this problem. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unicall and Private CID
Here is a log with level 255 when a Nextel phone tries to call in: Aug 2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 0001 [1/ 1/Idle /Idle ] Aug 2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Detected Aug 2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Making a new call with CRN 32769 Aug 2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1101 - [2/ 2/Idle /Idle ] Aug 2 15:38:18 WARNING[32670]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Detected Aug 2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 3 on [2/ 2/Seize ack /Seize ack] Aug 2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 on - [2/ 2/Seize ack /Seize ack] Aug 2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 3 off [2/ 2/Group A /DNIS request ] Aug 2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 off - [2/ 2/Group A /DNIS request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 7 on [2/ 2/Group A /DNIS request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 on - [2/ 2/Group A /DNIS request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 7 off [2/ 2/Group A /DNIS request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 off - [2/ 2/Group A /DNIS request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 2 on [2/ 2/Group A /DNIS request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 on - [2/ 2/Group A /DNIS request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 2 off [2/ 2/Group A /DNIS request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 off - [2/ 2/Group A /DNIS request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 8 on [2/ 2/Group A /DNIS request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 6 on - [2/ 2/Group A /DNIS request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 8 off [2/ 2/Group C /Category req ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 6 off - [2/ 2/Group C /Category req ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 2 on [2/ 2/Group C /Category req ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 on - [2/ 2/Group C /Category req ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 2 off [2/ 2/Group C /ANI request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 off - [2/ 2/Group C /ANI request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - F on [2/ 2/Group C /ANI request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 R2 prot. err. [2/ 2/Group C /ANI request ] cause 32772 - Unexpected MF6 signal Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Protocol failure -- Unicall/1 protocol error. Cause 32772 Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel echo cancel Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 - 0001 [1/ 1/Idle /Idle ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Detected Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Making a new call with CRN 32769 Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 1101 - [2/ 2/Idle /Idle ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:2644 handle_uc_event: Unicall/2 event Detected Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 - 3 on [2/ 2/Seize ack /Seize ack] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 1 on - [2/ 2/Seize ack /Seize ack
Re: [asterisk-users] Teliax Quality of Service
On 8/2/07, John Meksavan wrote: Asterisk Users, I recently ran into some problems with the quality of service with Teliax. This occurred on August 1, 2007 with a dropped outbound call, audio quality isse on the callee side- not hearing me well on callee side, and sending DTMF tones (configured for RFC2833). Am I the only Teliax customer having this problem? ditto here this week, random breaks in audio, garbled voice etc. My softphones dialing in from outside had no audio issues. Others on teliax forums suggested I switch to SIP since iax2 is aggressively evolving and teliax equipment is experiencing some incompatibilities with recent * iax releases. I changed codecs from gsm to ulaw, voice quality improved but same random breaks. It seems like when I am ready to go live with my Asterisk PBX System, I run into quality of service issues with the SIP provider. Consider having some fall back options from alternate providers since it doesn't cost a whole lot to keep an active account. Who should I go with that would guarantee me quality service just like an analog line? I have heard that there is no such thing unless your provider you have a dedicated, or at least highly reliable, circuit between the two of you : http://en.wikipedia.org/wiki/User_Datagram_Protocol UDP does not guarantee reliability or ordering in the way that TCP does. Datagrams may arrive out of order, appear duplicated, or go missing without notice. Avoiding the overhead of checking whether every packet actually arrived makes UDP faster and more efficient, at least for applications that do not need guaranteed delivery. Time-sensitive applications often use UDP because dropped packets are preferable to delayed packets... One of the reasons Time Warner, Armstrong, Cox and other cable broadband guys are able to offer fairly reliable voip service is that they control the pipes between their VoIP proxies and their end users. It is also the reason vonage, teliax and other 3rd party vendors have more issues. I used broadvox a few years ago, if the callee answered before the caller had heard a ring the line went dead :-) -baji. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI/T1 data rate...
Andre Courchesne - Consultant wrote: Hi all, First, this is not my first PRI/T1 Asterisk deployement. Did several with Bell, Telus, AllStream, Rogers but this is my first with Videotron. Just spoke with the person taking the order and on top of the standard settings (switch, coding,...) she asked me about data rate (56k or 64k). If she's selling a PRI and is asking this, then she is a bit clueless. 56k applies to RBS (CAS) T1s. http://www.cisco.com/en/US/products/sw/iosswrel/ps1826/products_feature_guide09186a0080080eb1.html Since I have never been asked this question before and can find anything relevant in the configuration files, I would like to have your inputs on this. Thanks, Andre Courchesne ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Andres Technical Support http://www.telesip.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A simple IVR extension problem
Hi list, I am running TDM11b + Asterisk-1.4.9 + Zaptel-1.4.4 + Libpri-1.4.1 on CentOS 5. I am having trouble to make my simple IVR extension work, here is relevant config: zapata.conf context=incoming signalling=fxs_ks channel = 4 context=internal signalling=fxo_ks channel = 1 - extensions.conf: [office] exten = s,1,Dial(Zap/1,30) [home] exten = s,1,Macro(stdexten,106,SIP/ht286,t) [incoming] ; incoming calls from the FXO port are directed to this context from zapata.conf exten = s,1,Answer exten = s,1,Background(enter-ext-of-person) exten = s,n,WaitExten(20) exten = 100,1,Dial(Zap/1,30) exten = 106,1,Macro(stdexten,106,SIP/ht286) exten = 101,1,Macro(stdexten,101,SIP/vli) exten = 107,1,AGI(math.agi) exten = 108,1,Playback(12) ;exten = s,1,GotoIfTime(9:00-16:30|mon-fri|*|*?office,s,1) ;exten = s,n,GotoIfTime(17:00-9:00|*|*|*?home,s,1) When I call my PSTN number, I can hear the enter-ext-of-person message, but once I press any one of the extension number, Asterisk sometime execute the relevant extension application, sometime not at all. If I comment the IVR extensions config and simply use : exten = s,1,GotoIfTime(9:00-16:30|mon-fri|*|*?office,s,1) exten = s,n,GotoIfTime(17:00-9:00|*|*|*?home,s,1) I can always get call My console message: ( Asterisk did not execute relevant extension in the last two call after I entered the extension digit) -- Starting simple switch on 'Zap/4-1' [Aug 2 13:46:38] NOTICE[4429]: chan_zap.c:6373 ss_thread: Got event 18 (Ring Begin)... [Aug 2 13:46:40] NOTICE[4429]: chan_zap.c:6373 ss_thread: Got event 2 (Ring/Answered)... -- Executing [EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(Zap/4-1, enter-ext-of-person) in new stack -- Zap/4-1 Playing 'enter-ext-of-person' (language 'en') -- Executing [EMAIL PROTECTED]:3] WaitExten(Zap/4-1, 20) in new stack == CDR updated on Zap/4-1 -- Executing [EMAIL PROTECTED]:1] Macro(Zap/4-1, stdexten|101|SIP/vli|t) in new stack -- Executing [EMAIL PROTECTED]:1] Dial(Zap/4-1, SIP/vli|20) in new stack -- Called vli -- SIP/vli-08353298 is ringing -- SIP/vli-08353298 answered Zap/4-1 == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'Zap/4-1' in macro 'stdexten' == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' [Aug 2 13:47:32] NOTICE[4437]: chan_zap.c:6373 ss_thread: Got event 18 (Ring Begin)... [Aug 2 13:47:33] ERROR[4437]: callerid.c:564 callerid_feed: fsk_serie made mylen 0 (-168) [Aug 2 13:47:33] WARNING[4437]: chan_zap.c:6405 ss_thread: CallerID feed failed: Success [Aug 2 13:47:33] WARNING[4437]: chan_zap.c:6505 ss_thread: CallerID returned with error on channel 'Zap/4-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(Zap/4-1, enter-ext-of-person) in new stack -- Zap/4-1 Playing 'enter-ext-of-person' (language 'en') -- Executing [EMAIL PROTECTED]:3] WaitExten(Zap/4-1, 20) in new stack == CDR updated on Zap/4-1 -- Executing [EMAIL PROTECTED]:1] Dial(Zap/4-1, Zap/1|30) in new stack -- Called 1 -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Zap/1-1 answered Zap/4-1 -- Native bridging Zap/4-1 and Zap/1-1 -- Hungup 'Zap/1-1' == Spawn extension (incoming, 100, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' [Aug 2 13:48:22] NOTICE[]: chan_zap.c:6373 ss_thread: Got event 18 (Ring Begin)... [Aug 2 13:48:23] ERROR[]: callerid.c:564 callerid_feed: fsk_serie made mylen 0 (-9) [Aug 2 13:48:23] WARNING[]: chan_zap.c:6405 ss_thread: CallerID feed failed: Success [Aug 2 13:48:23] WARNING[]: chan_zap.c:6505 ss_thread: CallerID returned with error on channel 'Zap/4-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(Zap/4-1, enter-ext-of-person) in new stack -- Zap/4-1 Playing 'enter-ext-of-person' (language 'en') -- Executing [EMAIL PROTECTED]:3] WaitExten(Zap/4-1, 20) in new stack == CDR updated on Zap/4-1 -- Executing [EMAIL PROTECTED]:1] AGI(Zap/4-1, math.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/math.agi -- Playing 'math-game-welcome' (escape_digits=) (sample_offset 0) -- Playing 'math-game-next' (escape_digits=) (sample_offset 0) -- Zap/4-1 Playing 'digits/17' (language 'en') -- Playing 'add' (escape_digits=) (sample_offset 0) -- Zap/4-1 Playing 'digits/15' (language 'en') -- Zap/4-1 Playing 'equals' (language 'en') -- Playing 'math-game-wrong' (escape_digits=) (sample_offset 0) -- Playing 'math-game-your-answer' (escape_digits=) (sample_offset 0) -- Zap/4-1 Playing 'digits/0' (language 'en') -- Playing 'math-game-right-answer' (escape_digits=) (sample_offset 0)
[asterisk-users] PhonicEQ T100P
Hi, Does anyone have any experience with the PhonixEQ T100P card? I wanted to know if it works fine with Asterisk without much of an issue. Thanks for your comments. TE100P 1 Port T1/E1 ISDN PRI Interface Card datasheet http://store.phoniceq.com/datasheet/te100p-datasheet.pdf TE100P offers unprecedented density and value in the telephony arena. Terminating one T1/E1 interface in a single PCI form-factor device, the TE100P harnesses the benefits of standard PC hardware and the open source Linux operating system. TE100P supports industry-standard telephony and data protocols, including both the CAS, CAS R2, Robbed Bit Signaling and Primary Rate ISDN protocol families for voice, as well as PPP, Cisco HDLC, and Frame Relay data modes. The card is fully supported by the Asterisk Open Source PBX and can drive both line-side and trunk-side interfaces, including supporting advanced call features. In addition to conventional telephony, Asterisk extends the strengths of the TE100P to provide Voice over IP and ultra low-latency TDM over Ethernet for greater efficiency and flexibility. Important Note: TE100P is for use with both a 3.3 *and* 5.0 volt PCI slot. Ritesh ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Agent Question
Dnia 2007-08-01, o godz. 11:47:42 Jason Adams [EMAIL PROTECTED] napisał(a): Hi, All, I have a question about agents and queues. Right now we have about 4 queues in our system. Some agents are in multiple queues. Our main queue is for technical support and it's by far our busiest queue as well. We have the autologoff feature set to 14 sec right now in the agents.conf file. The problem I'm running into is we don't want people in our sales queue (who are also in the support queue) to be auto logged off from the sales queue. Is there a good way to seperate agents and only have the them logged off from the support queue and not the sales queue? Make different agents with same channels and people behind the phone. Agent logged of is an agent logged off (in my understanding). -- .: Jakub Głazik, .: email jabber: zytekatnuxi.pl ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax Quality of Service
Asterisk Users, In my setup, I have a T1 service with McleodUSA and I am using the SIP protocol. I am considering switching back to analog lines because quality of service outweighs the cost savings at my work. Any good SIP providers out there? From: Baji Panchumarti [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Teliax Quality of Service Date: Thu, 2 Aug 2007 16:40:27 -0400 On 8/2/07, John Meksavan wrote: Asterisk Users, I recently ran into some problems with the quality of service with Teliax. This occurred on August 1, 2007 with a dropped outbound call, audio quality isse on the callee side- not hearing me well on callee side, and sending DTMF tones (configured for RFC2833). Am I the only Teliax customer having this problem? ditto here this week, random breaks in audio, garbled voice etc. My softphones dialing in from outside had no audio issues. Others on teliax forums suggested I switch to SIP since iax2 is aggressively evolving and teliax equipment is experiencing some incompatibilities with recent * iax releases. I changed codecs from gsm to ulaw, voice quality improved but same random breaks. It seems like when I am ready to go live with my Asterisk PBX System, I run into quality of service issues with the SIP provider. Consider having some fall back options from alternate providers since it doesn't cost a whole lot to keep an active account. Who should I go with that would guarantee me quality service just like an analog line? I have heard that there is no such thing unless your provider you have a dedicated, or at least highly reliable, circuit between the two of you : http://en.wikipedia.org/wiki/User_Datagram_Protocol UDP does not guarantee reliability or ordering in the way that TCP does. Datagrams may arrive out of order, appear duplicated, or go missing without notice. Avoiding the overhead of checking whether every packet actually arrived makes UDP faster and more efficient, at least for applications that do not need guaranteed delivery. Time-sensitive applications often use UDP because dropped packets are preferable to delayed packets... One of the reasons Time Warner, Armstrong, Cox and other cable broadband guys are able to offer fairly reliable voip service is that they control the pipes between their VoIP proxies and their end users. It is also the reason vonage, teliax and other 3rd party vendors have more issues. I used broadvox a few years ago, if the callee answered before the caller had heard a ring the line went dead :-) -baji. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Tease your brain--play Clink! Win cool prizes! http://club.live.com/clink.aspx?icid=clink_hotmailtextlink2 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax Quality of Service
Any good SIP providers out there? It really depends where you are. We're serving pretty much only Los Angeles and Seattle rather than the entire US, and thus by focusing our efforts on those limited markets we can achieve pretty good quality and reliability. Servers are 15 ms away, less potential for congestion, etc. Of course with the Internet being a best-effort network there are no guarantees, but by minimizing the potential for trouble you can achieve decent quality nevertheless. So, try to find a provider near you focusing on your market. Luki ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hints and Noop
Hello, I want to get rid of bunch of useless notices in the logs when the hint is not found, does setting the hint to noop for everything breaks anything? exten = _X.,hint,NoOp So far it did what I wanted. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI/T1 data rate...
Steve Totaro wrote: I knew someone would have an explanation that makes sense. I have NEVER done anything but PRI from the Telco. Wouldn't the question of signaling and switchtype negate the need to ask for data rate? Yes. But these are probably telco ordering droids, meaning that all they know is that they have to fill in the blanks. I recently ordered a LD PRI from a carrier. I wanted PRI, switchtype either 5ESS or preferrably National. The order got kicked because I didn't specify whether or not I wanted EM and which type of em (immediate, wink, etc) I wanted. I seem to recall a couple of other totally non-relevant questions that I had to specify as well... Or, more specifically, convince the droid which was checking the order for completeness that they weren't needed. -forrest ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pri call by call trunking?
Hi, Erik, Never heard of call-by-call trunking. Are you in Minnesota? What carrier are you using? --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Anderson Sent: Wednesday, August 01, 2007 12:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] pri call by call trunking? I've been working with a telco for the past two days trying to get a PRI span up and running. This is a small-ish telco and I get the feeling they don't do this very often. Anyway, they specified a pretty standard setup: ni2 switchtype, esf framing, b8zs coding, etc. All of my b-channels are up, but we're having a heck of a time getting the d-channel to come up. He finds out that this is an asterisk system and says that to get this working, I'm going to need to turn on call-by-call trunking. Have any of you heard of this? I certainly haven't. A quick google search doesn't turn up anything. Thoughts? This is a Sangoma A102 card, by the way. In this case, though, I don't think that's of any relevance. -Erik -- Erik Anderson http://andersonfam.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI/T1 data rate...
Thanks to all that responded so quickly. It was helpfull to me and I hope other that will be asked the same question by telcos. Andre Courchesne ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI/T1 data rate...
On Thu, 2 Aug 2007, Forrest W. Christian wrote: The order got kicked because I didn't specify whether or not I wanted EM and which type of em (immediate, wink, etc) I wanted. Are you serious? Which ILEC is this? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicall and Private CID
Hi Carlos, I suggest you download spandsp-0.0.3pre22. (http://www.neuwald.biz/files/spandsp-0.0.3pre22.gz) I don´t know why , spandsp after that uses digits 1,2..8,9,A,B,C,D,E,F instead of 1,2,..,9,0,A,B,C,D,E. So, do you get F digits that are incompatible with mfcr2 . Luis A P Barbosa. 2007/8/2, Carlos Chavez [EMAIL PROTECTED]: Here is a log with level 255 when a Nextel phone tries to call in: Aug 2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 0001 [1/ 1/Idle /Idle ] Aug 2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Detected Aug 2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Making a new call with CRN 32769 Aug 2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1101 - [2/ 2/Idle /Idle ] Aug 2 15:38:18 WARNING[32670]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Detected Aug 2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 3 on [2/ 2/Seize ack /Seize ack] Aug 2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 on - [2/ 2/Seize ack /Seize ack] Aug 2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 3 off [2/ 2/Group A /DNIS request ] Aug 2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 off - [2/ 2/Group A /DNIS request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 7 on [2/ 2/Group A /DNIS request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 on - [2/ 2/Group A /DNIS request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 7 off [2/ 2/Group A /DNIS request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 off - [2/ 2/Group A /DNIS request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 2 on [2/ 2/Group A /DNIS request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 on - [2/ 2/Group A /DNIS request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 2 off [2/ 2/Group A /DNIS request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 off - [2/ 2/Group A /DNIS request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 8 on [2/ 2/Group A /DNIS request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 6 on - [2/ 2/Group A /DNIS request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 8 off [2/ 2/Group C /Category req ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 6 off - [2/ 2/Group C /Category req ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 2 on [2/ 2/Group C /Category req ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 on - [2/ 2/Group C /Category req ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 2 off [2/ 2/Group C /ANI request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 off - [2/ 2/Group C /ANI request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - F on [2/ 2/Group C /ANI request ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 R2 prot. err. [2/ 2/Group C /ANI request ] cause 32772 - Unexpected MF6 signal Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Protocol failure -- Unicall/1 protocol error. Cause 32772 Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel echo cancel Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 - 0001 [1/ 1/Idle /Idle ] Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Detected Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Making a new call with CRN 32769 Aug 2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 1101 - [2/
Re: [asterisk-users] PRI/T1 data rate...
On 8/2/07, Forrest W. Christian [EMAIL PROTECTED] wrote: Steve Totaro wrote: I knew someone would have an explanation that makes sense. I have NEVER done anything but PRI from the Telco. Wouldn't the question of signaling and switchtype negate the need to ask for data rate? Yes. But these are probably telco ordering droids, meaning that all they know is that they have to fill in the blanks. I recently ordered a LD PRI from a carrier. I wanted PRI, switchtype either 5ESS or preferrably National. The order got kicked because I didn't specify whether or not I wanted EM and which type of em (immediate, wink, etc) I wanted. I seem to recall a couple of other totally non-relevant questions that I had to specify as well... Or, more specifically, convince the droid which was checking the order for completeness that they weren't needed. -forrest PLEASE tell me who that carrier is. I work with an inept company that doesn't even know what ANI and CPN mean. Well our ANI and CPN are one and the same. A bunch of inbred hicks somewhere in Alabama. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk configuration directly with Mandi (Speechphone)
Has anyone set up Speechphone (Mandi) directly with Asterisk and not used an ATA? If so, could you share how you did it? TIA ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pri call by call trunking?
On 8/1/07, Erik Anderson [EMAIL PROTECTED] wrote: I've been working with a telco for the past two days trying to get a PRI span up and running. This is a small-ish telco and I get the feeling they don't do this very often. Anyway, they specified a pretty standard setup: ni2 switchtype, esf framing, b8zs coding, etc. All of my b-channels are up, but we're having a heck of a time getting the d-channel to come up. He finds out that this is an asterisk system and says that to get this working, I'm going to need to turn on call-by-call trunking. Have any of you heard of this? I certainly haven't. A quick google search doesn't turn up anything. Thoughts? This is a Sangoma A102 card, by the way. In this case, though, I don't think that's of any relevance. Yes it is. Try setting TDMV_DCHAN = 0 in your wanpipe1.conf ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI - DS3 Calls Dropped
I think one critical aspect to explore here is, what exactly is meant by drop the DS3 service to redundant back-ups? SONET protection switching inside their transport core should not impact your DS3s or voice-bearing T1s if it is within the 50 ms tolerance of the standard. I imagine some SONET line equipment even jitter-buffers to account for this sort of thing, although that is speculation. Or is there another form of back-up involved? What is it? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI - DS3 Calls Dropped
I have a customer installation with an Adtran DS3 mux. The DS1's go into my Asterisk servers that run IVR/Call recorders. The DS3 provider is Qwest, and they tell me that they routinely drop the DS3 service to redundant back-up's and that this is a common practice that happens thousands of times to DS3 lines daily across the US without any service interruptions. They say that the switchover time is covered by specification, AND my Adtran mux. I've modified the zaptel drivers to mask out hardware alarms (alarmdebounce) with no success. I think that I have weeded out all poss ability of alarm conditions coming from the physical T1 framers, and I still get #6 and #8 HDLC PRI errors that are dropping all calls on the servers. I've been working on this for about 6 months now, and I can't be live that all asterisk PRI installations are dropping all calls at a rate of 2 per month like I am. Everything appears to be in spec. The carrier gave me several example installations that are not dropping calls but they are not using Asterisk. They all were using my Adtran mux though. I have 3 Adtran muxes and have tried them all. I have used different T1 cards. The results are all the same. 2 times a month, all of the calls are dropped. I get a page when this happens, and then some times another from my customer who gets an e-mail about a call that was lost. I'm starting to think that this installation is unique in its size and that this is just a byproduct of free software, but I never see this behavior in any smaller installations. I think that my best bet is that someone here knows how to manipulate the q931 to not drop the calls or somehow stop the errors from dropping the calls. Thanks, Bob ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI - DS3 Calls Dropped
On Thu, 2 Aug 2007, [EMAIL PROTECTED] wrote: That is the back-up we are talking about here, the call loss is %100 when this happens. Ah, I see. So, if I understand you correctly, what you appear to be saying is that somewhere between your Asterisk box and your Adtran mux this is not being handled gracefully. From where do you have the impression it would be any different if you had a different M13 mux or weren't using a PC-based open-source TDM platform? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI - DS3 Calls Dropped
I'm battling from a position here where I don't have a different DS3 to play with, and I don't have a differnet mux. I'm being leaned on completely with the argument that everyone else does this without any service interruptions. I'm asking this group for the secret. I have run out of arguments about the service and the switchover. Like I said, I've been looking at this for 6 months now. Alex Balashov [EMAIL PROTECTED] wrote: On Thu, 2 Aug 2007, [EMAIL PROTECTED] wrote: That is the back-up we are talking about here, the call loss is %100 when this happens. Ah, I see. So, if I understand you correctly, what you appear to be saying is that somewhere between your Asterisk box and your Adtran mux this is not being handled gracefully. From where do you have the impression it would be any different if you had a different M13 mux or weren't using a PC-based open-source TDM platform? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI - DS3 Calls Dropped
That is the back-up we are talking about here, the call loss is %100 when this happens. Alex Balashov [EMAIL PROTECTED] wrote: I think one critical aspect to explore here is, what exactly is meant by drop the DS3 service to redundant back-ups? SONET protection switching inside their transport core should not impact your DS3s or voice-bearing T1s if it is within the 50 ms tolerance of the standard. I imagine some SONET line equipment even jitter-buffers to account for this sort of thing, although that is speculation. Or is there another form of back-up involved? What is it? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Re: PRI - DS3 Calls Dropped
That is exacly what is happening. The 50ms interruption is disturbing everything up to the chan_zap level, even though I have supressed the yellow alarms. Date: Thu, 2 Aug 2007 23:58:11 -0400 (EDT) From: Alex Balashov [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PRI - DS3 Calls Dropped I think one critical aspect to explore here is, what exactly is meant by drop the DS3 service to redundant back-ups? SONET protection switching inside their transport core should not impact your DS3s or voice-bearing T1s if it is within the 50 ms tolerance of the standard. I imagine some SONET line equipment even jitter-buffers to account for this sort of thing, although that is speculation. Or is there another form of back-up involved? What is it? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI - DS3 Calls Dropped
Can someone here tell me why a switchover at the SONET level CAN disturb my DS1? From the beginning, I though that carrier and messages were contained in this specification. Alex Balashov [EMAIL PROTECTED] wrote: On Thu, 2 Aug 2007, [EMAIL PROTECTED] wrote: That is the back-up we are talking about here, the call loss is %100 when this happens. Ah, I see. So, if I understand you correctly, what you appear to be saying is that somewhere between your Asterisk box and your Adtran mux this is not being handled gracefully. From where do you have the impression it would be any different if you had a different M13 mux or weren't using a PC-based open-source TDM platform? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk1.2 to 1.4 g711a fax
marek cervenka wrote: hi, i have problem with pass-through faxing with this scenario hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.2.X(xen virtual) - linksys ATA i can fax to fax2mail on hylafax but after upgrade asterisk2 to 1.4 faxing is not working hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.4.X(xen virtual) - linksys ATA configuration is same do you hava any idea what is changed in 1.4 in g711 pass-through faxing? thanks Jitterbuffer behavior, maybe? Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dtmf get data
The problems started when we weren't getting the dtmf codes properly. ie. if we type 4000500600 we werent getting the dtmf digits as is , we were getting wrong dtmf codes like 405600 something to that effect. This was without any changes to the system. We later moved the asterisk server to another machine, where everything worked fine except for the problem i mentioned. After you mail, i check the versions of the software running. Nothing had changed on the asterisk-java side. But there was a change in the asterisk version. The one where previous m/c had 1.2.10 and the current m/c had 1.2.23. I have switched back to 1.2.10 and it seems to work fine !!! Thanks a lot. My mistake not to have checked the versions. Any idea why we were getting jumbled up dtmf codes in the first place. Nothing had changed on that system. We checked with the DID service provider and they said that network latency could be one of the reasons Best Regards, Shivram U On 8/3/07, Steve Totaro [EMAIL PROTECTED] wrote: Shivaram U wrote: Greetings, We have a handlewelcome.agi script which handles every new caller. For every new call we play a welcome message and ask the caller to enter a four digit code .. something on the lines Welcome... please enter the four digit number Our asterisk java agi script calls a function getData() with parameters such as the gsm file to play the message and the number of dtmf characters to receive. (the getData() call maps to the asterisk cmd get data) We have noticed that if a user keyed in the four digit code while the message is being played. the dtmf char received is only one. i.e if i key in 1007 before the message is fully played we get only 1 char. but if i wait for the whole message to complete then there is no problem we receive the complete code 1007 . But i remember that before we could enter and receive the dtmf digits correctly even if the message was playing. Best Regards, Shivram Before what? An upgrade? A change in code? The answer to that question may reveal a bug or just an answer to your question. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A simple IVR extension problem
Might want to start by proving out your DTMF by just sending the calls to something like VoiceMailMain(). When going into the voicemail system, see if you can reliably get DTMF to work while entering mailbox numbers and password and moving around the VM system.. At first glance it sure sounds to me like a DTMF issue of some sort. Regards, Todd R. -- Prestige Messaging Live Answering Services SIP or Toll-Free Connectivity Light Accounts From $14.95/mo http://www.PrestigeMessaging.com On 8/2/07, Vincent Li [EMAIL PROTECTED] wrote: Hi list, I am running TDM11b + Asterisk-1.4.9 + Zaptel-1.4.4 + Libpri-1.4.1 on CentOS 5. I am having trouble to make my simple IVR extension work, here is relevant config: zapata.conf context=incoming signalling=fxs_ks channel = 4 context=internal signalling=fxo_ks channel = 1 - extensions.conf: [office] exten = s,1,Dial(Zap/1,30) [home] exten = s,1,Macro(stdexten,106,SIP/ht286,t) [incoming] ; incoming calls from the FXO port are directed to this context from zapata.conf exten = s,1,Answer exten = s,1,Background(enter-ext-of-person) exten = s,n,WaitExten(20) exten = 100,1,Dial(Zap/1,30) exten = 106,1,Macro(stdexten,106,SIP/ht286) exten = 101,1,Macro(stdexten,101,SIP/vli) exten = 107,1,AGI(math.agi) exten = 108,1,Playback(12) ;exten = s,1,GotoIfTime(9:00-16:30|mon-fri|*|*?office,s,1) ;exten = s,n,GotoIfTime(17:00-9:00|*|*|*?home,s,1) When I call my PSTN number, I can hear the enter-ext-of-person message, but once I press any one of the extension number, Asterisk sometime execute the relevant extension application, sometime not at all. If I comment the IVR extensions config and simply use : exten = s,1,GotoIfTime(9:00-16:30|mon-fri|*|*?office,s,1) exten = s,n,GotoIfTime(17:00-9:00|*|*|*?home,s,1) I can always get call My console message: ( Asterisk did not execute relevant extension in the last two call after I entered the extension digit) -- Starting simple switch on 'Zap/4-1' [Aug 2 13:46:38] NOTICE[4429]: chan_zap.c:6373 ss_thread: Got event 18 (Ring Begin)... [Aug 2 13:46:40] NOTICE[4429]: chan_zap.c:6373 ss_thread: Got event 2 (Ring/Answered)... -- Executing [EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(Zap/4-1, enter-ext-of-person) in new stack -- Zap/4-1 Playing 'enter-ext-of-person' (language 'en') -- Executing [ [EMAIL PROTECTED]:3] WaitExten(Zap/4-1, 20) in new stack == CDR updated on Zap/4-1 -- Executing [EMAIL PROTECTED]:1] Macro(Zap/4-1, stdexten|101|SIP/vli|t) in new stack -- Executing [EMAIL PROTECTED]:1] Dial(Zap/4-1, SIP/vli|20) in new stack -- Called vli -- SIP/vli-08353298 is ringing -- SIP/vli-08353298 answered Zap/4-1 == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'Zap/4-1' in macro 'stdexten' == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' [Aug 2 13:47:32] NOTICE[4437]: chan_zap.c:6373 ss_thread: Got event 18 (Ring Begin)... [Aug 2 13:47:33] ERROR[4437]: callerid.c:564 callerid_feed: fsk_serie made mylen 0 (-168) [Aug 2 13:47:33] WARNING[4437]: chan_zap.c:6405 ss_thread: CallerID feed failed: Success [Aug 2 13:47:33] WARNING[4437]: chan_zap.c:6505 ss_thread: CallerID returned with error on channel 'Zap/4-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(Zap/4-1, enter-ext-of-person) in new stack -- Zap/4-1 Playing 'enter-ext-of-person' (language 'en') -- Executing [EMAIL PROTECTED]:3] WaitExten(Zap/4-1, 20) in new stack == CDR updated on Zap/4-1 -- Executing [EMAIL PROTECTED]:1] Dial(Zap/4-1, Zap/1|30) in new stack -- Called 1 -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Zap/1-1 answered Zap/4-1 -- Native bridging Zap/4-1 and Zap/1-1 -- Hungup 'Zap/1-1' == Spawn extension (incoming, 100, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' [Aug 2 13:48:22] NOTICE[]: chan_zap.c:6373 ss_thread: Got event 18 (Ring Begin)... [Aug 2 13:48:23] ERROR[]: callerid.c :564 callerid_feed: fsk_serie made mylen 0 (-9) [Aug 2 13:48:23] WARNING[]: chan_zap.c:6405 ss_thread: CallerID feed failed: Success [Aug 2 13:48:23] WARNING[]: chan_zap.c:6505 ss_thread: CallerID returned with error on channel 'Zap/4-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(Zap/4-1, enter-ext-of-person) in new stack -- Zap/4-1 Playing 'enter-ext-of-person' (language 'en') -- Executing [EMAIL PROTECTED]:3] WaitExten(Zap/4-1, 20) in new stack == CDR updated on Zap/4-1 -- Executing [EMAIL PROTECTED]:1] AGI(Zap/4-1, math.agi) in new stack --
Re: [asterisk-users] PRI/T1 data rate...
Andrew Joakimsen wrote: PLEASE tell me who that carrier is. I work with an inept company that doesn't even know what ANI and CPN mean. Well our ANI and CPN are one and the same. A bunch of inbred hicks somewhere in Alabama. The underlying carrier is actually really clueful (Qwest the LD carrier, not Qwest the ILEC). I was really impressed with the provisioning tech who did a really nice job of running over the parameters which are tweakable, but often you don't get to tweak, like ANI vs CID delivery (ANI please), etc.. Mainly it was strictly an issue with undertrained sales people who probably aren't paid well enough to stick around long enough to get a clue. This circuit was actually purchased through one of the website low-price brokers, which then get prices from LD resellers which actually then buy in bulk from carriers. The broker fills out the form and submits it to the LD reseller who then reviews it (and in this case complains because all of the required questions not relevant to a PRI were skipped). This is actually pretty common. Generally once you can get through the nightmare of the order, things go well. In the weird, non-relevant question category, my favorite is the whole discussion I've had every time I've ordered a PRI from my local ILEC regarding the number of presented digits. I do realize that back in old pbx days, you wanted the telco to send you say 4 digits which corresponded to your extension number, and the question is still valid with PRI - athough why anyone would want less than the full 10 digit NANPA number is beyond me. Obviously it isn't as common as I think because my ordering process normally goes something like this: Q:How many presentation digits? Me: 10 Q:Really? Me: Yes, 10 digits. Q: Are you sure your switch can handle 10 digits? Me: Yes. It routes them to the correct extension and that way I don't have to worry about number conflicts. Q: Ohhhkay, (with that tone of voice of I'm not going to protect you from your own stupidity.). (pause) Q: Now what type of start would you like on these trunks? * *ok, the last question doesn't usually get asked by the ILEC, but I just *had* to add it for humor purposes... -forrest ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
