Re: [asterisk-users] 1.6, t.38 and zoiper - t38_udptl or t38pt_udptl ?
This should be sufficient to get it to work from zoiper to zoiper. http://asteriskguru.org/tutorials/zoiper2zoiperfaxt38.html If you would still experience any issues, please send us a packet capture + a description of the setup. Cheers and good luck! Zoa Olivier wrote: Hello, 2008/12/5 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] I will publish a tutorial in the beginning of next week about how to configure Zoiper and Asterisk to do t.38 together. Zoa. Where will you publish this tuto ? Regards ___ -- 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] CDR Design
Quote : Like I said earlier - the CDR's aren't reliable enough for a billing platform (as you've rightly pointed out) but are OK for very basic call logging (something the customer can look at). I couldn't disagree more. The CDRs is the MOST reliable source for billing purposes (in postpaid mode that is - for prepaid you have to use something else (although even then the CDRs can be helpful for consistency checks)). Other alternatives (e.g. radius) could not give you the same level of consistency as the CDRs (although better than other implementations because the gateway retries to send the packet many times before giving up). What would happen if your radius server was overloaded and could not process accounting packets for a few secs/mins/hours? What would happen if the network is down (and the event processing handler is in another box)? All these calls would be lost. This can rarely be seen with CDRs logging. Because, whatever might happen you can always count on them to rectify the situation. I think the same can be said about other event based billing setups. The gateway itself cannot (and shouldn't really) be aware if the event was successfully processed by the handling back-end so you end up with inconsistent data and lost calls. Now, a combination of the two (e.g. radius+CDRs) can cover all the possible gone-wrong scenarios. But in order for that to work you need all the detailing you can get in the CDR. If ,however, you don't want to load your gateway with complex CDRs you could entirely turn them off (or parts of it e.g. b-leg logging, log only a few details etc). Andrew Thomas wrote: Thanks for this Greyman - it's all beginning to make sense now ;). I agree that the 'loss of CDR upon txfr' is a nasty bug which does need to be addressed before anything else (assuming it hasn't been already). But, wouldn't it be better if you could ignore the CDR's completely and use an event based system? This would give you ALL the information you need. All that remains is to filter out the un-required bits. Like I said earlier - the CDR's aren't reliable enough for a billing platform (as you've rightly pointed out) but are OK for very basic call logging (something the customer can look at). Hopefully, the murf'ster will chirp in here :). Cheers Andy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Grey Man Sent: 05 December 2008 09:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CDR Design On Fri, Dec 5, 2008 at 8:26 AM, Andrew Thomas [EMAIL PROTECTED] wrote: In summary: Leave CDR exactly as it is and create a new CEL (Call Event Logging) module (optional in modules.conf) that puts out (and does not accept) call event information (ie. a one-way fire-and-forget output from Asterisk). Hi Andrew and Others, This thread is actually part of a discussion that has been going on for over a year. The links below provide the background to the whole thing. http://www.asterisk.org/node/48358 http://bugs.digium.com/view.php?id=11849 http://lists.digium.com/pipermail/asterisk-users/2008-January/204856.htm l Up until recently the approach was to try and fix the specific bugs with transfer CDRs as a typical bug. There is now a realisation that that is a lot trickier than inially thought so it's been decided to try and come up with a good design for the Asterisk CDR sub-system. Regards, Greyman. ___ -- 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
Re: [asterisk-users] 1.6, t.38 and zoiper - t38_udptl or t38pt_udptl ?
I will publish a tutorial in the beginning of next week about how to configure Zoiper and Asterisk to do t.38 together. (and while doing so test the latest version again to make sure it really works) Feel free to send us any bugtickets if you think something is broken, in the case of t.38 support, please also include sip and udtpl captures. Cheers, Zoa. Olivier wrote: 2008/12/5 Stefan Lekov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Olivier wrote: 2008/12/3 Olivier [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 2008/12/3 Steve Underwood [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Hi, I would be interested in any reports of anyone getting a T.38 FAX to send or receive successfully with Zoiper. I've tried to test my T.38 implementation against more than one revision of Zoiper, and I yet to see it behave sanely. I could receive yesterday from Zoiper 2.18 Windows free edition with ReceiveFAX installed with latest Asterisk 1.6.X. Would you like more details ? To be more precise, I only tried once and I wouldn't swear T.38 mode was used (I've checked) : maybe ReceiveFax would allow G711 inbound faxes. Steve Olivier wrote: 2008/12/3 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] I tried sending faxes through Zoiper (Zoiper to Zoiper) last week and the program crashed. After an update it stopped crashing but still could not send a FAX. I then tried Kapanga (the free version has a limited 30 days FAX sending capability) and it worked. This might be of little use No it has much value to me. I was already suspicious about Zoiper as I could send fax between other endpoints. Originally, I thought using the same software on both could help to validate other settings (tuning of sip.conf) but to my surprise I couldn't find much about Zoiper fax capabilities. I have the rough feeling Zoiper can't receive fax at the moment or either, tuning Asterisk to allow that is not simple. For sending, I won't rate it at the moment. to you since I used Freeswitch with t38 pass-trhough, but my point is : first find a client that supports t38 FAX capabilites that work and then try asterisk. Our company had success sending FAX using t38 through asterisk along time ago so I guess you will get there sooner or later. Olivier wrote: Hi, 1. Has anyone got any success when send a TIFF file form one zoiper softphone to another ? I tried using Zoiper 2.18 free edition in windows but I'm seeing 415 Unsupported media replies. 2. Here (http://www.voipinfo.org/wiki/view/Asterisk+T.38), you can read : Also, try using: t38_udptl=yes t38pt_rtp=no t38pt_tcp=no ... in the general section of the sip.conf and under the VoIP provider account as well as the fax account. But above, you can read [general] t38pt_udptl = yes Has this parameter name
[asterisk-users] remote phones, no audio to PSTN
Odd problem, where some remote phones, at users homes, dial and connect fine, no matter what the destination is. Bad phones, SIP to SIP, between remote and office, or remote to remote, work and have good audio, but no audio, at all, to PSTN or Cell phones.Phone can be moved to office and work fine. I'm perplexed, at this hour. OK, ok, at most hours. joe a. ___ -- 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] CDR Design
Billing and logging should not be confused theoretically - I agree. But in practice, the logging of the calls (not other events of the system) IS used for billing purposes. The start and finish time is not enough for many (I not that it is not enough for me). The accountcode is not enough for me either. From my CDRs I have to extract all the information about which provider tried-and-failed or tried-and-succeeded to terminate the call. So I need the terminator's info in the CDRs. This is the only way that I can monitor what my providers charge me (and believe me, never take for granted that your provider charge you with pre-agreed rates, mistakes happen :)). Also, having the terminator's data in the CDR is the only way that I can calculate metrics such as ASR, ACD, mean PDD etc. And I can't imagine taking all this info from a logging module that mixes CDR log events with other ones (hardware events, user agent registrations, etc.) Since there is no agreement on WHAT to log and since we have the option to put a lot of info in the CDRs I think the right way to do it is provide the capability of every single detail that COULD be logged and let the end user choose WHAT to log through the configuration. I cannot understand tha benefit of a minimal/fixed/non-flexible CDR logging capability when can have the flexibility to go from minimal to complex depending on a configuration entry in a proper configuration file. P.S. Sometimes I wonder if I am the only one in the VoIP world that finds terminator information in the CDRs useful (including failed calls). P.S. Sometime we use the term billing only for customer billing processes which nowadays is incorrect or insufficient. Billing in today's demanding VoIP business means : 1. Customer Billing : we all know what that is 2. Provider CDRs cross-check : as I said above, you have to know what your provider charges you in order to catch mistakes and in order to able to produce profit/loss reports. 3. QoS metrics : ASR, ACD, PDD to name a few. These cannot be calculated without proper termination info from the CDRs. I see LCR modules being introduced now and then in the asterisk community but they all seem a little useless if the above metrics cannot be extracted from the CDRs. What is the benefit of having a low cost provider in your LCR if its ASR equals to 0.0001 %? and how can you measure its ASR if the terminator's info (both failed and successful) is not in the CDRs? Andrew Thomas wrote: It seems to me that we are confusing billing and logging here. Call billing only really needs the start and finish (like we get now) - but proper call logging requires all steps. Do we leave CDR's as they are (for billing purposes) and have a separate 'event' driven log for call logging? Or do we change the CDR structure to accommodate logging as well? Personally, a separate 'event' log seems preferable to me as this keeps existing billing platforms useable. It just means the logging programs will need to be re-written to look at a new database for events. I know we have the AMI - but that puts out a lot more information than is needed for simple logging (and requires something to prune and store the events in a database of some sort). Any thoughts? Andrew Thomas Technical Services Manager DataVox Ltd Saddleworth Business Centre Huddersfield Road Delph, Oldham OL3 5DF ___ -- 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] 1.6, t.38 and zoiper - t38_udptl or t38pt_udptl ?
I tried sending faxes through Zoiper (Zoiper to Zoiper) last week and the program crashed. After an update it stopped crashing but still could not send a FAX. I then tried Kapanga (the free version has a limited 30 days FAX sending capability) and it worked. This might be of little use to you since I used Freeswitch with t38 pass-trhough, but my point is : first find a client that supports t38 FAX capabilites that work and then try asterisk. Our company had success sending FAX using t38 through asterisk along time ago so I guess you will get there sooner or later. Olivier wrote: Hi, 1. Has anyone got any success when send a TIFF file form one zoiper softphone to another ? I tried using Zoiper 2.18 free edition in windows but I'm seeing 415 Unsupported media replies. 2. Here (http://www.voipinfo.org/wiki/view/Asterisk+T.38), you can read : Also, try using: t38_udptl=yes t38pt_rtp=no t38pt_tcp=no ... in the general section of the sip.conf and under the VoIP provider account as well as the fax account. But above, you can read [general] t38pt_udptl = yes Has this parameter name changed between 1.4 to 1.6 from t38_udptl to t38pt_udptl ? A asterisk remains silent when I add an unknown parameter foo=bar, it would perfect if someone could point the right name (t38_udptl or t38pt_udptl). Regards ___ -- 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] asterisk ooh323 avaya (URGENT!!!)
I would go for chan_h323. Much more stable since 1.4 and the config more close to the other channel configs too. We used it on production for a long time and it worked well although a little heavy cpu-wise. To get started you need to install openh323 and pwlib from here http://sourceforge.net/project/showfiles.php?group_id=80674 and the ./configure and make menuselect will detect it and let you build it along with asterisk. Be careful with the paths when installing them though. And watch the output of the asterisk configure command for possible errors. David fire wrote: hi sorry about the urgent but it is urgent i have problems configuring a connection between asterisk and avaya using H323. the module i am usign is ooh323 what do you need to help me? and any tip or hint? thanks!!! David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] CDR Desgin
I agree with Freddi and would like to add that a field indicating the order of the outgoing legs would be very useful. For billing purposes one could benefit very much if one new the order of the providers that were called in a specific call. Freddi Hansen wrote: To me the obvious answer is to provide a CDR for every call leg so for A calling B via Asterisk there would be two CDRs produced. It's far far easier to disregard the unwanted CDRs than it is to try and generate the missing ones and in some cases it's virtually impossible. If it's weighed up I think people would vote to have accurate CDRs ahead of anything else and if single legs are the best way to do that then it's the way it should be done. In addition with single leg CDRs it will solve the dilemna about acommodating every possible call scenario that I know has caused you a lot of consternation over the last 18 months. Sure it's a change from the current situation so maybe needs to use the standard apporach of a configuration setting to opt in initially before becoming the default in the subsequent major release. OK, Greyman, your logic is solid. If we provide a CDR implementation that just generates the individual call legs, and ties them together via the linkedid (see team/group/newcdr), then both camps should be able to derive the info they need for billing, via hopefully not-overly-complex SQL queries to a backend db. I'll modify my RFC to reflect this line of thinking. Yes, it is a bit of shift. And, yes, the implementation will make this new approach optional, and not default. But, pardon if I make it available via the CEL domain come implementation time. It should take me a week to rehash my document; perhaps longer (I'm in bugfix mode, and this borderline development work); in the meantime, those with decided CDR needs might make their wishes known, if they do not think this approach will work. Speak now, or forever hold your peace; or forever complain... or whatever. If this is particularly distressing to you, perhaps your fears might be slightly assuaged when you read the details... I was part of a team that did design a multiservice billing system about 15 years ago and the solution people seems to agree on here (and me to) looks pretty much the same i.e one call may consist of several calls legs. In addition to the linkedid it would be nice to have an indication in the cdr that tells us that 'this is the lastone on this linked id'. Our experience was that we shouldn't for load reasons work with cdr's in the immidiate multileg format in the DB. So we did collect cdr's in a tmp DB until we got the the record with end marker set. We would then produce our final cdr for the actual service, store it in billing col. and delete it from the multileg col. When a new service is created we only have to make a the new customized cdr, we don't have to touch the generation of the multileg format. Freddi ___ -- 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] CDR Design
Yes, I know we are suggesting the same thing... I just thought you are suggesting putting this multidimensional CDR in one row (which of course requires data structure other than a simple comma separated row - XML perhaps). I did not understand you were referring to a conceptual multi-dimensional and not an actual multidimensional storage method. Anthony Francis wrote: We are suggesting the same thing, what you describe is multidimensional. If you think of the cdr's as being in a database, and say you wanted to have it show you all the calls today and all the calls that are associated with that call. Your select grabs the first dimension, a list of all calls. Then using the unique identifier of each call you build a second dimension of the related calls. [EMAIL PROTECTED] wrote: In order to avoid a multidimensional schema we could have 1 cdr per call leg. So , for instance, one call that had 3 different dial() commands as outgoing attempts would be described by 4 CDRs (1 for the incoming leg that has all the originating channel data and 3 for the outgoing legs that hold all the terminating channel's data). Those CDRs would be bound by a unique identifier field (the same for all 4). The terminating CDRs could be also identified by a increment field that indicates the order that the channels were called. Another issue is that failed attempts should also be logged because this is valuable info for many (or at least have the option to choose the desired behavior - which is available in asterisk as we speak). Anthony Francis wrote: It is my belief that before redesigning the CDR engine some time should be spent looking at current PSTN CDR formats and what information is kept in them. The main problem that I feel we face is that calls can be complicated, but we want the record of it to not be. In reality a CDR that incorporates all information about a call would have at least two dimensions. In the first you would have the base call record as we do now, in the second we would have the event list. The event list would be a time indexed list of event names and attributes, just as you would currently store event information. The event list would be your glue (with a bit of front end logic of course.) that would relate a call that dialed X external numbers to the X different new CDR's that generated. That would allow you all the call path info you could ever want. The most important thing would be a new config file that allows an administrator granular control over what information is important to them. And of course a keep it simple stupid mode that just writes the top level cdr as it does now. [EMAIL PROTECTED] wrote: I think that the custom cdr back-end can be successfully used to maximize or minimize the CDRs detailing on a per-needs basis. Furthermore, the CDR() function gives plenty of room for even more detailing. In my opinion the detail level (fields) is not the issue with the CDRs generation nor is the lack of backends (asterisk gives a lot of different backends to store your CDRs). I find the issue with asterisk CDRs to be in the lack of proper CDRs generation for the B-leg of every call. If we want to really track what happens during a call through the CDRs one has to have all the details not only for the incoming channel but for the outgoing one as well. Furthermore, one needs to be able to tweak the B-leg CDRs like he does with the incoming legs. So what needs to be done in my opinion is record every B-leg CDR when such an event occurs and give the user access to the CDR info by extending the CDR() function (so that one can specify the channel of the CDR that is being tweaked) or creating a seperate one for the outgoing channels. Grey Man wrote: I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal (link ommitted to see if email gets accepted). After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation that is already overly so. I think it's a mistake to try and think about all the different call scenarios and come up with little tricks for the more complicated ones. There will always be something missed; app_shotgun initiates calls to 100 random numbers and as soon as three or more calls are answered it will start randonly transferring them amongst each other at 2 second intervals. I think it's important to clarify at the outset what a CDR should be. The most fundamental requirement for CDRs is that they accurately record the following pieces of information for EVERY call entering or leaving the system (note every means every and not; channel calls but not peer calls). 1. Destination (aka as A Number) 2
Re: [asterisk-users] CDR Design
I think that the custom cdr back-end can be successfully used to maximize or minimize the CDRs detailing on a per-needs basis. Furthermore, the CDR() function gives plenty of room for even more detailing. In my opinion the detail level (fields) is not the issue with the CDRs generation nor is the lack of backends (asterisk gives a lot of different backends to store your CDRs). I find the issue with asterisk CDRs to be in the lack of proper CDRs generation for the B-leg of every call. If we want to really track what happens during a call through the CDRs one has to have all the details not only for the incoming channel but for the outgoing one as well. Furthermore, one needs to be able to tweak the B-leg CDRs like he does with the incoming legs. So what needs to be done in my opinion is record every B-leg CDR when such an event occurs and give the user access to the CDR info by extending the CDR() function (so that one can specify the channel of the CDR that is being tweaked) or creating a seperate one for the outgoing channels. Grey Man wrote: I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal (link ommitted to see if email gets accepted). After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation that is already overly so. I think it's a mistake to try and think about all the different call scenarios and come up with little tricks for the more complicated ones. There will always be something missed; app_shotgun initiates calls to 100 random numbers and as soon as three or more calls are answered it will start randonly transferring them amongst each other at 2 second intervals. I think it's important to clarify at the outset what a CDR should be. The most fundamental requirement for CDRs is that they accurately record the following pieces of information for EVERY call entering or leaving the system (note every means every and not; channel calls but not peer calls). 1. Destination (aka as A Number) 2. AccountCode (aka as B Number) 3. Call Start Time (answer time), 4. Duration. Of course adding extra information can be very useful and I'm not proposing any fields be removed from the current implementation (although for pity's sake one change that should be made it to use a GUID/UUID for the CDR's uniqueid and save endless confusion). People that really do need verbose or enhanced CDRs to do things like tracking a call's flow as it travels in and out of queues, parking lots etc. would be better off using AMI or the new CEL and not CDRs. At the very least if problems arise with their call flow tracking they will still be able to rely on the accuracy of the CDRs to piece it altogether to work out what's going wrong. My proposal of creating a 1-to-1 relationship between CDRs and Asterisk channels already exsits but somewhere along the line it's going awry. As an experiment, and to actually do something instead of continually moaning about it, I started commenting out the blocks of code in res_featrures.c and sip_channel.c that muck around with the channel CDRs when a transfer occurs. The results of that were that the CDRs for blind and attended transfers actually got better! They're still not quite right but are pretty close with only one CDR on each having a wrong destination. Regards, Greyman. ___ -- 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] CDR Design
In order to avoid a multidimensional schema we could have 1 cdr per call leg. So , for instance, one call that had 3 different dial() commands as outgoing attempts would be described by 4 CDRs (1 for the incoming leg that has all the originating channel data and 3 for the outgoing legs that hold all the terminating channel's data). Those CDRs would be bound by a unique identifier field (the same for all 4). The terminating CDRs could be also identified by a increment field that indicates the order that the channels were called. Another issue is that failed attempts should also be logged because this is valuable info for many (or at least have the option to choose the desired behavior - which is available in asterisk as we speak). Anthony Francis wrote: It is my belief that before redesigning the CDR engine some time should be spent looking at current PSTN CDR formats and what information is kept in them. The main problem that I feel we face is that calls can be complicated, but we want the record of it to not be. In reality a CDR that incorporates all information about a call would have at least two dimensions. In the first you would have the base call record as we do now, in the second we would have the event list. The event list would be a time indexed list of event names and attributes, just as you would currently store event information. The event list would be your glue (with a bit of front end logic of course.) that would relate a call that dialed X external numbers to the X different new CDR's that generated. That would allow you all the call path info you could ever want. The most important thing would be a new config file that allows an administrator granular control over what information is important to them. And of course a keep it simple stupid mode that just writes the top level cdr as it does now. [EMAIL PROTECTED] wrote: I think that the custom cdr back-end can be successfully used to maximize or minimize the CDRs detailing on a per-needs basis. Furthermore, the CDR() function gives plenty of room for even more detailing. In my opinion the detail level (fields) is not the issue with the CDRs generation nor is the lack of backends (asterisk gives a lot of different backends to store your CDRs). I find the issue with asterisk CDRs to be in the lack of proper CDRs generation for the B-leg of every call. If we want to really track what happens during a call through the CDRs one has to have all the details not only for the incoming channel but for the outgoing one as well. Furthermore, one needs to be able to tweak the B-leg CDRs like he does with the incoming legs. So what needs to be done in my opinion is record every B-leg CDR when such an event occurs and give the user access to the CDR info by extending the CDR() function (so that one can specify the channel of the CDR that is being tweaked) or creating a seperate one for the outgoing channels. Grey Man wrote: I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal (link ommitted to see if email gets accepted). After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation that is already overly so. I think it's a mistake to try and think about all the different call scenarios and come up with little tricks for the more complicated ones. There will always be something missed; app_shotgun initiates calls to 100 random numbers and as soon as three or more calls are answered it will start randonly transferring them amongst each other at 2 second intervals. I think it's important to clarify at the outset what a CDR should be. The most fundamental requirement for CDRs is that they accurately record the following pieces of information for EVERY call entering or leaving the system (note every means every and not; channel calls but not peer calls). 1. Destination (aka as A Number) 2. AccountCode (aka as B Number) 3. Call Start Time (answer time), 4. Duration. Of course adding extra information can be very useful and I'm not proposing any fields be removed from the current implementation (although for pity's sake one change that should be made it to use a GUID/UUID for the CDR's uniqueid and save endless confusion). People that really do need verbose or enhanced CDRs to do things like tracking a call's flow as it travels in and out of queues, parking lots etc. would be better off using AMI or the new CEL and not CDRs. At the very least if problems arise with their call flow tracking they will still be able to rely on the accuracy of the CDRs to piece it altogether to work out what's going wrong. My proposal of creating a 1-to-1 relationship between CDRs and Asterisk channels already exsits but somewhere along the line
Re: [asterisk-users] CDR Desgin
If we only implement A-D cdr we lose information. On the other hand, if we implement all 3 CDRs for one call we can either use this info or ignore it and act like its not there. The first way is prohibiting for some users. The second one can match any scenario with none to little effort. Steve Murphy wrote: On Sat, 2008-11-22 at 04:02 +, Grey Man wrote: I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal: http://svn.digium.com/svn/asterisk/team/murf/RFCs. After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation that is already overly so. I think it's a mistake to try and think about all the different call scenarios and come up with little tricks for the more complicated ones. There will always be something missed; app_shotgun initiates calls to 100 random numbers and as soon as three or more calls are answered it will start randonly transferring them amongst each other at 2 second intervals. I think it's important to clarify at the outset what a CDR should be. The most fundamental requirement for CDRs is that they accurately record the following pieces of information for EVERY call entering or leaving the system (note every means every and not; channel calls but not peer calls). 1. Destination (aka as A Number) 2. AccountCode (aka as B Number) 3. Call Start Time (answer time), 4. Duration. Of course adding extra information can be very useful and I'm not proposing any fields be removed from the current implementation (although for pity's sake one change that should be made it to use a GUID/UUID for the CDR's uniqueid and save endless confusion). People that really do need verbose or enhanced CDRs to do things like tracking a call's flow as it travels in and out of queues, parking lots etc. would be better off using AMI or the new CEL and not CDRs. At the very least if problems arise with their call flow tracking they will still be able to rely on the accuracy of the CDRs to piece it altogether to work out what's going wrong. My proposal of creating a 1-to-1 relationship between CDRs and Asterisk channels already exsits but somewhere along the line it's going awry. As an experiment, and to actually do something instead of continually moaning about it, I started commenting out the blocks of code in res_featrures.c and sip_channel.c that muck around with the channel CDRs when a transfer occurs. The results of that were that the CDRs for blind and attended transfers actually got better! They're still not quite right but are pretty close with only one CDR on each having a wrong detstination. Regards, Greyman. Greyman-- For the moment, let's not worry about the implementation. Let's get consensus on the spec first. In the scenario, where A calls B, B xfers A to C, C xfers A to D, or some such similar scenario, half the world wants a single CDR for A, from the time it started, to the time it hung up with D. The other half wants A-B's dial and bridge, a cdr for A C's bridge, a CDR for A D's bridge, and mayhaps some CDRs to describe the xfers, where B xfers A to C and C xfers A to D. My document is pointing in the former direction, and either we need to spec both, or pick one. murf ___ -- 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] How does IMAP notify Asterisk that I've read amessage?
On 11/22/2008 at 11:17 PM, Barry L. Kline [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I have an Asterisk box sitting between the PSTN and a legacy PBX. I have successfully configured Asterisk to use IMAP for voicemail and have written the necessary script to turn the MWI indicator (via a .call file to the PBX) on and off. I have two issues still outstanding: 1) When the user listens to his voice mail via the phone, it will be announced that the caller is unknown, in spite of the fact that the email headers show the appropriate Callerid(num) information. I can live with that, but I'll eventually need to get it fixed. 2) If I listen to the voicemail using my email client, the MWI on the phone is not turned off, which isn't surprising given that my script needs to be called to generate the .call file. What I don't know is how, exactly, Asterisk is notified that I've listened to my voicemail via email. Does Asterisk poll the server? If so, where is the frequency of the poll set? Can Asterisk be configured to call the script again when the messages are read and the MWI should be turned off? The docs don't say anything about this and I've not found anything in my googling that has given me any leads? I'm currently using Asterisk 1.4.22. Thanks for any information that you can provide. Barry Regarding #2 - There is nothing in Asterisk, at this time, that is able to check the status of a attachment in a message in an external system. AFAIK. Sending an email is one thing. Being able to check the status of an attachment in a specific message, in any one of a variety of systems, is, I think, asking too much. I don't think most email systems keep track of wether or not an attachment has been read, in any case. joe a. ___ -- 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] P2P
Hello List, i would like to set up the following concept: Scenario 1: = VOIP-Phone -tcp/udp- VOIP-Phone (direct P2P between two phones. Those phones have be he hard phones. No Software such as KPhone or something) Scenario 2: = VOIP-Phone -tcp/udp- Asterisk -tcp/udp- VOIP-Phone (Those phones also have be he hard phones.) Are this scenarios possible? What hardware do i need for this? Has anyone any recommendations? I guess for Scenario 2 the Asterisk box just need a simple pc with a network card? Thanks, Mario ___ -- 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] P2P
Hi Valetin, Valentin Bud wrote: Are the VoIP phone mobile on the internet or in fixed locations? If they are in fixed locations and they have to go through internet to reach the asterisk box, the way *i* would do it is with VPN tunnels. If they are in the same location (LAN) it is very simple, you just need the phones and an asterisk box with a network card as you said. You configure the phones to register with the asterisk and configure the dialplan and you are good to go. They are in the same network/lan. Can you recommend and hard phones for this task? Are there phones which can be used without asterisk in between them? Thanks, Mario ___ -- 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] Looking for a good lightweight Linux softPhone
Hello, I'm the person responsable for the zoiper roadmap, comments inline snip This all started because Zoiper really annoyed me - they keep sending me beta versions of their software (which is nice, thanks you), and they keep on compiling it for ubuntu or some other distribution of linux I don't use and dynamically link it with libraries I don't have. I emailled them months ago about these issues - that it won't work with the current Debian Stable/Etch disty, and after several weeks they tell me they're not going to support it. We are definately going to support it, i know it takes a bit longer than normal, because we are insanely busy with some huge projects, but it's definately coming ! It's really pissed me off because idefisk was small, clean and light-weight and ran under all my systems. Now they tell me zoiper is going to have video and who knows what else in it. I feel it's bloated out of all proportion, just like Ekiga. Even with video and all the other functionality its still going to be small, and light weight, the size will increase with a few hundred kb only. So yes, I could compile up asterisk for my workstation, my laptop and who knows what else, but I don't want to! I used idefisk for a long time, then they turned it into zoiper, so I struggled with that, but it was never the same. There were sound compatability issues, and now Linux distribution issues - they apear to compile it under some bleeding-edge Ubuntu distro and the binary won't run under Debian stable. I made the mistake of asking the dev team to make it for ubuntu although im a big debian fan myself. I thought it would work on most of the desktop based distros that typically have newer libs. But i did not expect it not to work on debian, the actual work to make it run on debian again is not that much, we just need a little time to install another build environment. My goal is for it to run on even more platforms than ever before. (more news on that in the next week). I just want to get back to basics and have a small application that I can drive using the keyboard that does nothing more than make and take calls using IAX. 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 ___ -- 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] directrtpsetup without reinvite
Hi, I want to be able to bridge two sip channels using direct RTP between my endpoints (Audio IP : not local) but without using reinvites. So I set up my asterisk sip endpoints as follows: [test1] type=friend host=dynamic username=test1 dtmfmode=info context=test_rtp allow=all canreinvite=no directrtpsetup=yes [test2] type=friend host=dynamic username=test2 dtmfmode=info context=test_rtp allow=all canreinvite=no directrtpsetup=yes ... but it doesn't work. How can I ensure that the RTP is not going through my asterisk box and that the re-invite method is not used? P.S. Both endpoints are using the same codec, so no codec translation takes place. ___ -- 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] directrtpsetup without reinvite
Yes, but my conf is quite straightforward, isn't it? No NAT etc... I just want to know what is the combination of directives that I have to use in order to achieve my goal. Is there going to be any support in the future for this feature? Because from the little I' ve seen in the mailing lists there is quite a lot of demand for it. It could sky-rocket asterisk as the first choice at the voip carrier level. By the way, I am using asterisk 1.4.22. Kevin P. Fleming wrote: Kristian Kielhofner wrote: What version of Asterisk is this? Last I heard (from Olle) this option was very experimental and should not be used on production systems. He even helpfully documented it that way in the sip.conf.sample file, along with a list of (known) cases where it will fail, although there are probably plenty more. ___ -- 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] Cisco 776M Any good for connection to local Asterisk server?
I'm pretty sure it is a VERY OLD ISDN router and you can't use it with *. The voice ports have no VoIP capabilities, they are just used directly from the ISDN line. Ronny Julian wrote: I found this at a local sale. I need to find a power supply for it. Before I do I wonder if anyone can tell me if it is any good for Asterisk? Looks to have 4 Ethernet ports and two phone ports. I did get the Cisco serial cable and some documentation. Also will this work with most any Cisco power supply? I see they all share the connector. Thanks! Ronny Julian ___ -- 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] say load new
Hello all, I would like to use say.conf settings but every time i restart asterisk i have to load manualy say load new is there a way to do it automaticaly i use asterisk 1.4.19 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] SIP: difference between Grandstream and Cisco when behind NAT
You generally don't need to enter the public IP of the router into the Cisco, just setting nat_enable to 1 is almost always sufficient. * is smart enough to realize that the IP of the packet is the public IP of the phone. Tony Mountifield wrote: I have used Grandstream phones for years, and have just started testing a Cisco 7940 (with SIP firmware 7.4). I have found something puzzling and don't know whether it's just a limitation or something I haven't done correctly. The Asterisk server is directly on the Internet with a public IP. The phones are on a private LAN with a NAT router to the Internet. The sip.conf entries for both phones say nat=yes. For the Grandstream, this is always sufficient to make it work properly with Asterisk, even though in the Grandstream config I have NAT traversal: no and leave Use NAT IP blank. All the clever stuff is done automatically by Asterisk. However, with the Cisco, that doesn't seem to be the case. I have found it necessary in the SIPDefault.cnf file to set nat_enable: 1 and then specify as nat_address the public address of my router. Is this normal? What is different between the Grandstream and the Cisco? Is there any way to avoid having to program the external address into the Cisco when it is behind NAT? Thanks in advance for any advice. Cheers Tony ___ -- 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] MOH Bad
I am running 1.4.10.1 and I am getting garbled MOH from calls within the same LAN with no firewall. Calls sound fine, but every 5-10 seconds the MOH gets garbled. I am using the stock MOH files. Any ideas where/how this could occur? There is no debug showing any issue with MOH. Thanks. Peder ___ -- 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] cisco phones getting SIP 401 unauthorized
What did the firewall change from and to? Did you have NAT enabled in * AND on the Cisco phones? FYI, if you have NAT enabled in both places, it will work if you have NAT in your setup or not. If you don't have it enabled in both places, then it may or may not work depending on your setup. Matt Gibson wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis Sent: Wednesday, October 08, 2008 10:13 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] cisco phones getting SIP 401 unauthorized Hi Jerry, Hm, okay. We had to use md5secret (instead of secret) in the sip.conf for our 7970's to get them to successfully register with asterisk. However, if you had them working before then I doubt this is the issue. You can try anyway though, http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+md5secret We use both secret= and md5secret= with the same password in each, one encrypted and one not encrypted - this seemed to let our 7970 register. Matt, I looked at this and did what it says for 1 of my phones. Still no go. I have a mix of polycom and cisco phones at this location and the polycom continue to work. The cisco are now having issues. The only change that had been made is the customer changed their firewall. All addresses remained the same just a new firewall. Polycom works cisco is not registering. Any thoughts? Jerry Hi Jerry, Hmm. We had to replace our router with one that supported SIP ALG (we went cisco). However, since you mention all the phones are in the LAN this shouldn't make a difference. Does the problem go away if you go back to the old firewall? Thanks, Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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] cisco phones getting SIP 401 unauthorized
As a followup to my previous email, change nat_enable to 1 and reboot the phones. Jerry Geis wrote: Did you check sip.conf to make sure that the port is correctly set to 5060? Please show the output of Cli sip show peer peernumber and the contents of your SEPMAC.cnf file. Dave sip.conf has : bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=X.X.X.X; ress to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls One extension cisco in sip.conf is: [402] type=friend dtmfmode=rfc2833 username=402 secret=XXX disallow=all allow=ulaw allow=alaw host=dynamic context=local-sip nat=yes canreinvite=no callerid=John Smith 402 sip show peer 402 * Name : 402 Secret : Set MD5Secret: Not set Context : smvoice-sip Subscr.Cont. : Not set Language : AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : Yes Callerid : John Smith 402 Expire : -1 Insecure : no Nat : Always ACL : No CanReinvite : No PromiscRedir : No User=Phone : No Trust RPID : No Send RPID: No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : (Unspecified) Port 0 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 402 SIP Options : (none) Codecs : 0xc (ulaw|alaw) Codec Order : (ulaw,alaw) Status : Unmonitored Useragent: Reg. Contact : SIP Config file: # SIP Configuration Generic File (start) # Proxy Server proxy1_address: X.X.X.X proxy2_address: X.X.X.X proxy3_address: X.X.X.X proxy4_address: X.X.X.X proxy5_address: X.X.X.X proxy6_address: X.X.X.X # Line 1 Settings line1_name: 402 ; Line 1 Extension\User ID line1_displayname: 402 ; Line 1 Display Name line1_authname: 402 ; Line 1 Registration Authentication line1_password: 402 ; Line 1 Registration Password # Line 2 Settings line2_name: 403 ; Line 2 Extension\User ID line2_displayname: 403 ; Line 2 Display Name line2_authname: 403 ; Line 2 Registration Authentication line2_password: 403 ; Line 2 Registration Password # Emergency Proxy info proxy_emergency: proxy_emergency_port: 5060 # Backup Proxy info proxy_backup: proxy_backup_port: 5060 # Outbound Proxy info outbound_proxy: outbound_proxy_port: 5060 # NAT/Firewall Traversal nat_enable: 0 nat_address: voip_control_port: 5060 start_media_port: 16384 end_media_port: 32766 nat_received_processing: 1 # Phone Label (Text desired to be displayed in upper right corner) phone_label: JDA 402 ; Has no effect on SIP messaging # Time Zone phone will reside in time_zone: EST # Telnet Level (enable or disable the ability to telnet into this phone telnet_level: 0 ; 0-Disabled (default), 1-Enabled, 2-Privileged # Phone prompt/password for telnet/console session phone_prompt: Go Away ; Telnet/Console Prompt phone_password: cisco ; Telnet/Console Password proxy_register: 1 # Enable_VAD (1-enabled, 0-disabled) enable_vad: 0 # Network Media Type (auto, full100, full10, half100, half10) network_media_type: auto user_info: phone # URL for external Directory location #logo_url: http://10.0.1.3/10-20logo.bmp;; URL for branding logo to be used on phone display # SIP Configuration Generic File (stop) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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] MWI with Siemens Gigaset S450IP
Kevin P. Fleming wrote: Olivier wrote: 2. R Hook-flash key is now available to transfer calls. In s450IP web management server, its defaults settings are : Application-type: dtmf-relay Application-signal: 16 Is there anything to configure in features.conf, extensionsconf or elsewhere to trigger transfers when R key is pressed ? I don't believe there is any support for hook-flash style transfers over SIP in Asterisk; that key should be programmed to use standard SIP transfer methods, not DTMF emulation methods. do you have a suggestion, there is only two fields that can be filled in that to refer to the R key, Application-type: I think this is content type Application-signal: what it sends? Thanks for your help Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Dropping SIP support?
They are probably referring to the fact that the base 7960 is End of Life and the 7960G is probably going to be EOL soon as well, so they won't offer new firmware at the EOL milestone. They have been replaced by the 7961. Completely different firmware and configuration, but there still is support for SIP. Stefan Gofferje wrote: Michael Graves schrieb: Earlier today I glanced at Junction Networks blog and was surprised to find a post indicating that Cisco was dropping SIP support in their 79xx series phones. Here's t link: http://www.junctionnetworks.com/blog/charlotte/2008/09/19/junction-netwo rks-lab-cisco-7960-phones Is this true? What are they thinking? Only SCCP? AFAIK the other way around is true. Cisco is dropping SCCP. The new firmware is for SIP only but it's with some Cisco extensions as the latest CCMs are using SIP as preferred protocol. Could be that Cisco drops the standard SIP FW though. Terve, Stefan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Cisco 1841 - Can it be made SIP aware?
I've had the same experience. I probably have 20-30 customers with multiple SIP phones behind PIX running 6.3(5) (which has been out almost 3 years) and I have no issues at all. You can even have two phones behind a PIX being PAT'd to a single external IP with reinvite enabled in * and you will still get 2 way audio. The SIP Fixup makes changes inside the SIP packet for internal IPs. The nice thing is that you don't need to enable NAT on the remote * server either. It thinks the device is not behind NAT. I have customers with 20 phones behind one IP connecting to a remote * box with no issues at all and no special PIX config. Now the IOS firewall, that is a completely different animal and works completely different than the PIX/ASA. Stefan Gofferje wrote: Kristian Kielhofner schrieb: IMNSHO, the less SIP aware the better... I have to disable SIP inspection on every IOS/PIX device I come across. Fix the one-way audio problems on your proxy, registrar, etc (in the case, Asterisk). Most SIP ALGs are broken. Interesting. I have my Asterisk with RFC-1918 IPs behid a NATting PIX and the FIXUP SIP of the PIX makes it very easy for me to use my * as server for external clients as well as as client for SIP providers. The PIX nicely replaces the RFC-1918 IP in the SIP-traffic with the current (dynamic) public IP of itself and keeps track of the RTP traffic. Actually, it also chages the ports in the RTP negotiation and then automatically forward the RTP traffic to the ports, the * was offering. Very very convenient. If the IOS firewall in the newer routers make problems, maybe I should not change to an ISR as I planned :). Terve, Stefan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zap channel DTMF regeneration
I've got a problem that I hope someone here can shed some light on. It seems that in any calls going over Zap channels (either with a FXO card or PRI card), inbound audio is constantly monitored for DTMF tones, and then these tones are regenerated back in the audio stream either within * if the endpoint is set for inband dtmfmode, or sent out-of-band for the endpoint to regenerate. I have two applications where this has disasterous results: 1. Connecting to another system that does a lot of DTMF signaling 2. Trying to use an iaxmodem and hylafax to receive faxes. The DTMF detection code easily falses and at least mutes the audio. I realize there are certain things (like the IVR) that require the DTMF detection; but on calls from the zap channel to an endpoint, I can't have it muting the audio, and potentially regenerating the tones. Perhaps this could be configurable per extension entry? I think I recall seeing something that looked for the fax initial tone, and would at least disable any echo cancellation - does that also disable DTMF regeneration? I appreciate any help or pointers! Joe Click to go wireless with your computer, ultra fast speed. http://thirdpartyoffers.netzero.net/TGL2231/fc/Ioyw6ijmWbww9f2hhX4f2TcPxLtgcLJ4DlAvmab8VmG43fIiATJTRp/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reproduce DeadAGI behavior with AGI
Hi, I decided to migrate my scripts from DeadAGI to AGI (FastAGI). The no-exit-on-hangup behavior suited me just fine with DeadAGI. How can I make my AGI scripts (which are executed on another AGI server) NOT to exit when a hangup is detected? I used AGISIGHUP=no before calling the AGI script but it didn't work. I am using asterisk-java 0.3.1. on asterisk 1.4.21.1. Regards, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] H.323 -dtmf-
Hi All, would Asterisk 'transcode' H.245 alphanumeric DTMFs to an H.245 signal / rfc2833 H.323 device over G.729 codec ? Thanks for supporting, .TF ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie alert: VoIP hardware
A couple of years ago I started my Asterisk carrier with selling x100p cards and I think I sold around 100 of them in total to people who could actually contact me and new who I was. Yes, it is a poor man solution but at least it is a solution. And for the poor man it is the only thing available. The serious cards are not an option because they simply don't have the money to buy Digium, Sangoma or other professional stuff. I would never advice to use it in production situations but for testing and trying or when you simply don't have the money and want to become an Asterisk expert starting with an old PC and a X100P card it is good that these card is around. With some trying with the gains and the settings an x100p card can work pretty well. I have never got a card send back to me because the buyer was dissatisfied with the sound quality. I had a couple of dead on deliveries that where solved by sending an other card. A couple of times I had to assist with the settings to get it right. People are not stupid, they understand that a card priced EUR 20,- can't offer the same quaility as a card that cost EUR 100,- or even more. The same counts for the hfc isdn2 card. It certainly isn't perfect but for around ER 15,- you can't expect perfection. But it works, you can connect your isdn2 cable ad have inbound and outbound calls. I used the card on my own home asterisk for a long time. You can even connect 2 asterisk servers when setting the cards in different modes. For practicing and trying that is perfect because lots of people can not effort to spend EUR 1.500,- for two E1 cards and do the same trying with an ISDN30 connection between two boxes. I'm not advertising the use of this cards in production but we all should be glad that this cards are around and people who want to learn how to use and configure Asterisk can start with an old pc and a x100p card or an hfc isdn2 card. Starting with not the best hardware available is better then not starting because you have the impression that that is only possible after spending serious money on equipment. Erik de Wild Tripple-o Your Asterisk migration partner the Netherlands ___ -- 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] help with rotating number plan
An option to rotate between numbers is to add a queue to the system and add and as agents and pick the proper strategy (rrmemory or leastrecent). This has some advantages: - the calls are devided as you have in mind - when there are more calls coming in they are queued instead of a busy tone - you can scale by just adding an agent to the queue see http://www.voip-info.org/wiki-Asterisk+call+queues for further info Erik de Wild Tripple-o Your Asterisk migration partner I'm trying to come up with a quick, easy solution to have a static inbound number in my dialplan, rotate calling 2 numbers. Example: 1st call into asterisk exten = 1234,1,Dial(sip/,10) exten = 1234,n,Dial(sip/,10) 2nd call into asterisk exten = 1234,1,Dial(sip/,10) exten = 1234,n,Dial(sip/,10) We're kind off looking to do load balancing via the dial plan. But I'm having a little trouble getting the logic to trace 1st call in, 2nd call in, 1st call in, 2nd call in, etc. ___ -- 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] Dialplan, Extensions, etc. Worksheet
extension example. Only the reception can call the voiceprompt routine
exten = 6000/5000,n,Goto(recording,s,1) ; calling extension 6000
exten = 6001,1,Answer() ; just to
test the proper working of music on hold
exten = 6001,n,MusicOnHold()
exten = 6001,n,Hangup()
#include extensions.d/*.conf ; some additional .conf are in /
etc/asterisk/extensions.d/ just to keep the oversight of the dialplan
configuration
;
without ending up with one large extensions.conf
[plaza_inbound]
;;;
; numbers don't exist anymore
;;;
exten = 0307114197,1,Answer() ;
incoming line on number 0307114197
exten = 0307114197,n,Goto(inbound_menu,s,1) ; jump to menu
[inbound_menu]
exten = s,1,BackGround(plaza/menu) ; press 1 for reception, 2
fr registration department , 3 for info department
; or
for for the fiancial department
;reception
exten = 1,1,Dial(SIP/5000,10,t)
exten = 1,n,Playback(plaza/no_answer)
exten = 1,n,Hangup()
;registration department
exten = 2,1,Dial(SIP/
5001,10,t) ; phone
5001 rings for 10 seconds
exten = 2,n,Dial(SIP/
5000,10,t) ; phone
5000 (reception) rings 10 seconden
exten = 2,n,Playback(plaza/external_transfer) ;
exten = 2,n,Dial(IAX2/[EMAIL PROTECTED]/0621831234,10,t) ; using
iax2 trunk OOO50608 external number 0621831234 is called for 10 seconds
exten =
2
,n
,VoiceMail(5001) ;
if no phone answered the caller can leave a message in mailbox 5001
exten = 2,n,Hangup()
;information department
exten = 3,1,Dial(SIP/
5002,10,r) ; phone rings
for 10 seconds;
exten = 3,2,Dial(SIP/
5000,10,t) ; phone of
reception rings for 10 seconds
exten = 3,3,Dial(IAX2/[EMAIL PROTECTED]/0621832345,10,t) sing iax2
trunk OOO50608 external number 0621832345 is called for 10 seconds
exten =
3,4
,VoiceMail(5002) ;
no phone answered the callee can leave a message in mailbox 5002
exten = 3,5,Hangup()
;;
; finance department
;
exten =
4,1
,Voicemail(5003) ;
the finance guys/girl only communicate by voicemail
exten = 4,2,Hangup()
; i = invalid
exten = i,1,Playback(plaza/
invallid_input); when a digit other the 1,
2, 3 or 4 is entered the input is invalid so the invalid message is
played
exten = i,2,Goto,(s,
1) ;
and then the ibound call returns back to the menu
; t = time-out
exten = t,1,Playback(plaza/
goodbye); caller waited to long
exten = t,
3
,Hangup
;
not very customer friendly but the line hangs up
[plaza_outbound_nl
] ;
only national outbound calls
exten = _0Z.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},20,rt) ; iax2
trunk OOO50608 calls out, be aware of the numbermatching _0Z.
exten = _0Z.,
2
,Hangup
()
; Z
=[123456789]
[plaza_outbound_int]
exten = _00Z.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},20,rt) ;
number match _00X. allows international calls
exten = _00Z.,2,Hangup()
[plaza_no_autorisation]
exten = _X.,1,Playback(plaza/no_autorisation) ; in case you enter a
number that doesn't fit any other available extension (depends on what
is included)
exten = _X.,n,Hangup
/etc/asterisk/extension.d/plaza_voiceprompts.conf
;;;
; Erik de Wild
; quick and dirty routine for recording Media Plaza demo voiceprompts
; 26-05-2007
;;;
[recording]
exten = s,1,Read(VOICEPROMPT_NUMMER|plaza/voiceprompt_nummer|2|
noanswer|1|15); enter the voiceprompt number
exten = s,n,Read(OPNEMEN_AFLUISTEREN|plaza/opnemen_afluisteren|1|
noanswer|1|15) ; OPNEMEN_AFLUSTEREN=LISTEN_RECORD
; press
1 for recording or 2 for listening
exten = s,n,Macro(bestandsnaam,$
{VOICEPROMPT_NUMMER
Re: [asterisk-users] Disable transfer on all calls
Dinesh Nair пишет:
On Tue, 22 Apr 2008 11:54:41 +0100, Grey Man wrote:
The best option is to put a SIP Proxy in front of your Asterisk sever
and block REFER requests.
or just comment out the block in chan_sip.c which handles the refers.
Thanks to your answers, but i found more beautiful way to do this -
there is some system variable __TRANSFER_CONTEXT, which defines context
to handle the transfered number, so you can create a new context and
there you can do anything with transfered call - i just hang it up.
It's really strange that this is in fact undocumented function - you can
find it only in comments on wiki at voip-info.org. Man there said that
he found this variable while hacking source code of asterisk:
$ grep -R TRANSFER_CONTEXT /usr/src/asterisk-1.2.15/
/usr/src/asterisk-1.2.15/channels/chan_sip.c: *transfercontext =
pbx_builtin_getvar_helper(sip_pvt-owner, TRANSFER_CONTEXT);
/usr/src/asterisk-1.2.15/doc/README.variables:${TRANSFER_CONTEXT}
Context for transferred calls
/usr/src/asterisk-1.2.15/ChangeLog: * channels/chan_sip.c: chan_sip did
not use the TRANSFER_CONTEXT
/usr/src/asterisk-1.2.15/res/res_features.c: if
(!(transferer_real_context = pbx_builtin_getvar_helper(transferee,
TRANSFER_CONTEXT))
/usr/src/asterisk-1.2.15/res/res_features.c: !(transferer_real_context =
pbx_builtin_getvar_helper(transferer, TRANSFER_CONTEXT))) {
/usr/src/asterisk-1.2.15/res/res_features.c: if
(!(transferer_real_context=pbx_builtin_getvar_helper(transferee,
TRANSFER_CONTEXT))
/usr/src/asterisk-1.2.15/res/res_features.c:
!(transferer_real_context=pbx_builtin_getvar_helper(transferer,
TRANSFER_CONTEXT))) {
___
-- 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] Disable transfer on all calls
Hi folks, I have some asterisk 1.2 box with self-made billing, and I need to disable call transfer on all calls and directions. I turned it off in features.conf and there is no 'tT' option in all my Dial() commands, but users still able to transfer call using transfer function in ip of softphones (AFAIK this function uses SIP method REFER), so this transfers are hard to trace in CDR and my users can make a free call using trick with transfer:) I've googled it, but didn't find anything about my problem :( Thanks, Danila ___ -- 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] ----www.cdsportal.net---- wholesale voipprovider --starting at 1.1 cent per min
Better yet. They claim SIX 9's uptime :) On Mar 21, 2008, at 1:37 PM, Joshua Kinard wrote: Piling on... InterNIC says the domain was created almost a week ago, and expires in a year. The registrar is GoDaddy. The owner of the site is located in the Dominican Republic: C/1ra #15 Costa Criolla, Km9 Carr. Sanchez Santo Domingo, New York 0 Dominican Republic Registered through: GoDaddy.com, Inc. (http://www.godaddy.com) Domain Name: CDSPORTAL.NET Created on: 14-Mar-08 Expires on: 15-Mar-09 Last Updated on: 14-Mar-08 Administrative Contact: Almonte, Juan [EMAIL PROTECTED] JHALMONTE C/1ra #15 Costa Criolla, Km9 Carr. Sanchez Santo Domingo, New York 0 Dominican Republic (809) 220-3278 Judging by the site's purported function, it's nothing more than a front for telemarketers, autodialers, and other ilk of the telephony industry to annoy normal people with. How can you claim five 9's uptime when your domain isn't barely over a week old? Well, I guess if the system hasn't crashed within that first week. But that's hardly a valid measurement, unless you're comparing against Windows Millenium systems. I call scam. --J -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ]On Behalf Of Gonzalo Servat Sent: Friday, March 21, 2008 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] www.cdsportal.net wholesale voipprovider --starting at 1.1 cent per min I think this type of abuse is well deserved due to the way he intended to advertise his business, so I'll add a bit of wood to the fire. How about the sign-up form?? Some serious HTML design work going on there. - Gonzalo On Fri, Mar 21, 2008 at 1:15 PM, Tim Nelson [EMAIL PROTECTED] wrote: The template website, page titles, and Gmail contact address surely aren't very convincing. Another crappy VoIP reseller that will fail in a few months taking a handful of customers down... assuming they're legit to begin with. --Tim - Original Message - From: Outback Dingo [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 21, 2008 11:06:31 AM (GMT-0600) America/Chicago Subject: Re: [asterisk-users] www.cdsportal.net wholesale voip provider --starting at 1.1 cent per min My first thought looking at the site was SCAM!!! maybe my second thought would be SCRAM ... is this company even legit On Fri, Mar 21, 2008 at 10:35 PM, Tim Nelson [EMAIL PROTECTED] wrote: Apparently the list description of Non-commercial Discussion isn't clear enough. And now the obligatory beat down: Instant Emergency Response and Delay Free Connection... WOW! I don't even have to call for support because when I have an emergency, response is INSTANT. On top of that... they've also figured out how to eliminate latency!!! Super duper! But wait, theres more!!! They are interconnected with major US carriers like QUEST!!! Not to be confused with QWEST... the little telco company that misspells it's name to differentiate itself from the ULTRA MEGA HUGE telco QUEST. /sarcasm Tim Nelson Systems/Network Support Rockbochs Inc. - Original Message - From: Ignacio Ortega A. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion Asterisk-Users@lists.digium.com Sent: Friday, March 21, 2008 10:20:17 AM (GMT-0600) America/Chicago Subject: [asterisk-users] www.cdsportal.net wholesale voip provider --starting at 1.1 cent per min starting a 1.1 cent per min, rates may be better depending volume technical support we support all codecs using SIP / IAX2 predictive dialers, call centers and telemarketers are allowed free test account. if you have any question just contact us [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 ___ -- 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
Re: [asterisk-users] Post call QoS in Asterisk 1.4
I have absolutely no idea since I was not even aware of it. However, this may give you some hints as to where you can find more information: http://www.mail-archive.com/[EMAIL PROTECTED]/msg27124.html - Waldo On Feb 22, 2008, at 5:08 PM, Douglas Garstang wrote: It's time to ask this question again. Maybe I will get a reply one day. :) Asterisk 1.4 has some channel variables that you can inspect after a call is complete that will give you QoS metrics. Stuff like average round trip time, etc. Since there's only one set of variables, and calls will have two channels, which channel is this information for? Is it for one of the channels? Is it an aggregate of both channels? Who added this code and what where they thinking when they wrote it? Thanks, Doug. Never miss a thing. Make Yahoo your homepage. ___ -- 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] [Copfilter] Copy of quarantined email - *** SPAM *** [6.0/6.0] IAX2 client asked to authenticate against wrong
Problem: When I have more than one IAX2 connection (on server zuiderven), I have problems in receiving calls from IAX peers except for the first in the list as seen by the iax2 show peers command. In my tests it showed that by removing one by one the entries from the iax.conf file in the order as they were showed. It tried to authenticate to the next. Eventually after removing all but the groetstraat it finally worked for this peer. While tracing the information with iax2 set debug, I had the impression that the receiving asterisk server told the one that tried to set up the call in the AUTHREQ package which username to use to authenticate in the challenge. This server ofcourse does not know how to do that on the wrong username. Below is configuration information as well as a little iax2 debug information. My question is, what is missing in the iax2 configuration that this is happening. This problem started when I added the groetstraat configuration. TIA, Hans Feringa zuiderven asterisk = 1.4.18 (compiled from source) groetstraat asterisk = 1.4.10 (ubuntu repository) This is the local (zuiderven) iax.conf: register = **:[EMAIL PROTECTED] register = 8*:[EMAIL PROTECTED] register = 8*:[EMAIL PROTECTED] [groetstraat] type=friend context=groetstraat-in host=dynamic trunk=no qualify=yes secret= disallow=all allow=ulaw allow=alaw [iaxfwd] type=user context=iaxfwd auth=rsa inkeys=freeworlddialup disallow=all allow=ulaw allow=alaw allow=gsm allow=ilbc allow=g726 [iaxfwd] type=peer host=iax2.fwd.net username=* secret=*** qualify=yes disallow=all allow=ulaw allow=alaw allow=gsm allow=ilbc allow=g726 [ordina-pc] type=friend context=home host=dynamic nat=yes qualify=yes username=* secret= disallow=all allow=ulaw allow=alaw And this is the remote (groetstraat) iax.conf: [general] autokill=yes externip=8x.x.x.x jitterbuffer=no forcejitterbuffer=no tos=ef register = **:[EMAIL PROTECTED] [zuiderven] type=friend context=zuiderven-in host=dynamic trunk=no qualify=yes secret=*** deny=0.0.0.0/0.0.0.0 permit=8x.x.x.x/255.255.255.255 disallow=all allow=ulaw allow=alaw allow=gsm zuiderven: asterisk*CLI iax2 show peers Name/UsernameHost Mask Port Status ordina-pc/* (Unspecified) (D) 255.255.255.255 0 UNKNOWN iaxfwd/8*(Unspecified) (S) 255.255.255.255 4569 UNKNOWN groetstraat **.**.**.** (D) 255.255.255.255 4569 OK (26 ms) 3 iax2 peers [1 online, 2 offline, 0 unmonitored] Call from groetstraat results in: [Feb 9 08:51:07] NOTICE[11030]: chan_iax2.c:7761 socket_process: Host **.**.**.** failed to authenticate as ordina-pc This is not the peer it should authenticate as. Debugging iax2, I get the impression that the receiving server tells the remote asterisk to authenticate against this wrong name. Ofcourse it does not know how to, and the call fails. In the packet from te receiving asterisk server I see: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00016ms SCall: 2 DCall: 0 [groetstraat-ip:4569] VERSION : 2 CALLED NUMBER : 3815 CODEC_PREFS : (ulaw|alaw) CALLING NUMBER : 087875 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: asterisk LANGUAGE: nl FORMAT : 4 CAPABILITY : 57356 ADSICPE : 2 DATE TIME : 2008-02-09 09:34:18 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 7ms SCall: 1 DCall: 2 [groetstraat-ip:4569] AUTHMETHODS : 3 CHALLENGE : 208379767 USERNAME: ordina-pc asterisk*CLI Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00039ms SCall: 2 DCall: 1 [groetstraat-ip:4569] MD5 RESULT : 57ac54c7782a8db29baff75086a07dfb [Feb 9 09:36:44] NOTICE[11030]: chan_iax2.c:7761 socket_process: Host groetstraat-ip failed to authenticate as ordina-pc Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00039ms SCall: 1 DCall: 2 [groetstraat-ip:4569] Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REJECT Timestamp: 00024ms SCall: 1 DCall: 2 [groetstraat-ip:4569] CAUSE : No authority found CAUSE CODE : 50 asterisk*CLI Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00024ms SCall: 2 DCall: 1 [groetstraat-ip:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 00014ms SCall: 3 DCall: 0 [groetstraat-ip:4569] USERNAME: groetstraat REFRESH : 60 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGACK Timestamp: 00018ms SCall: 7 DCall: 3 [groetstraat-ip:4569] USERNAME: groetstraat DATE TIME : 2008-02-09 09
Re: [asterisk-users] resync linksys SPA9XX config file from Asterisk
I would like to know as well, it has never worked for me. On Dec 18, 2007 4:27 PM, JR Richardson [EMAIL PROTECTED] wrote: Hi All, Anyone know the sip header to send to a Linksys to resync it's config file? Thanks. JR -- JR Richardson Engineering for the Masses ___ --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] Graceful Asterisk Shutdown
You can also do
asterisk -rx stop gracefully
From any sort of script / crontab, etc.
On Dec 10, 2007 10:36 AM, Jeng Yu [EMAIL PROTECTED] wrote:
Thanks, All! And thanks, Oquendo! I will experiment
with this suggestion. I was actually thinking in terms
of a situation where it would be done
non-interactively.
Jeng
--- J. Oquendo [EMAIL PROTECTED] wrote:
Jeng Yu wrote:
This would be the ultimate graceful shutdown;
perfect
for routine system maintenance tasks on production
servers handling continuous traffic.
if [ `asterisk -rx show channels verbose|awk
'/active calls/{print
$1}'` -eq 0 ]
then asterisk -rx stop now
fi
--
J. Oquendo
SGFA #579 (FW+VPN v4.1)
SGFE #574 (FW+VPN v4.1)
I hear much of people's calling out to punish the
guilty, but very few are concerned to clear the
innocent. Daniel Defoe
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0xF684C42E
___
--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
__
Sent from Yahoo! Mail - a smarter inbox http://uk.mail.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
Re: [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.
What would you be reinventing? Asterisk can already get its configuration from a MySQL database. You could even add extra fields in that case to store the phone model and macaddress and integrate that into your own provisioning tools. Your application would then retrieve the configuration from the Asterisk SIP database, parse it and generate configurations for your phones. Very straight forward. The other option is create your own database with your own schema and then design your parser to create the asterisk configuration files and phone configuration files. This method has the advantage of not requiring any changes to your asterisk configuration. On Dec 7, 2007 9:12 PM, Philip Prindeville [EMAIL PROTECTED] wrote: That's sort of my point: that you have to reinvent it, and it's easy to get wrong. ___ --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] Open Asterisk Exchange Project
On Dec 7, 2007 11:28 AM, Michael Munger [EMAIL PROTECTED] wrote: Is there anyone interested in developing an open source Asterisk / MS Exchange solution? Please explain. This sounds interesting. But why MS exchange only? I think it's safe to say with good IMAP and LDAP support we can integrate with just about any decent enterprise messaging system. Think of all the Scalixes and Zimbras of the 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
Re: [asterisk-users] Asterisk SIP Microsoft Outlook Integration
OutCall is very confusing. The user has multiple options Call Contact the outlook option remains present (and does not work). That is confusing. You need a TAPI Driver for the easiest user experience. This plugs in to the Windows/Outlook framework. Not only does it work with the factory Outlook options any application that uses TAPI for placing calls will work with no added configuration/modification. There are various 3rd party TAPI drivers but I think these little things are items that need to be added to the Asterisk development. On Dec 5, 2007 12:26 PM, Jared Smith [EMAIL PROTECTED] wrote: On Wed, 2007-12-05 at 11:45 -0500, Michael Melia Jr. wrote: Does anyone know how I could integrate my Asterisk setup with Outlook One of the more popular ones seems to be Outcall, which is now open-source and available from http://outcall.sourceforge.net. I haven't tried it personally, so your mileage may vary. -- 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 ___ --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] Print CALLERID in CLI during pri debug
What don't you tell us what you are ultimately trying to do. You want the callerid next to the connect message in debug output... why? What will that help you to accomplish? On Dec 7, 2007 4:42 PM, Arpit Mehta [EMAIL PROTECTED] wrote: Ok so the call reference is the 'cr' field (q931.c) and how do I retrieve the caller id from this call reference ? On Dec 7, 2007 4:29 AM, Richard Revels [EMAIL PROTECTED] wrote: When the call sets up the 'call reference' is assigned. It will be unique for the duration of the call and other messages, like Connect, will reference it. At the same time, the setup will have indicated the caller ID info. Sent from my iPhone On Dec 6, 2007, at 10:28 PM, Arpit Mehta [EMAIL PROTECTED] wrote: Or in other words is there a way to map which message is from which CallerID ? On Dec 6, 2007 6:40 PM, Arpit Mehta [EMAIL PROTECTED] wrote: Hi all, I was wondering if it is possible to print the callerid value in the CLI when doing 'pri debug span 1' For example Call Ref: len= 2 (reference 2707/0xA93) (Terminator) Message type: CONNECT (7) [18 03 a9 83 97] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 23 ] [1e 02 81 82] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) I would like to print '1234567890 Message type: CONNECT (7) ... ... ' where 1234567890 is the callerid Thanks Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --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 -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --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] G729 on wrong bus
OkWhat is the issue? Does your G729 not work? Anyways who cares about the CPU? If you have a 32 bit Linux you need a 32 bit program. ___ --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] Door Intercom? (was: Re: Phone with public address functionality)
Besides grandstream-doorphone transplant surgery, no. But it does have PoE. It's cheap, especially if you already have a doorphone. If you used a GXP-2000 you can use the display and it supports XML idle screens. On Dec 4, 2007 2:53 AM, Nick Seraphin [EMAIL PROTECTED] wrote: On a similar note... has anyone ever seen a SIP-based door intercom unit? Functionality I'm looking for is... basically an outdoor rated weather resistant speaker with 1 button and microphone, when the button is pressed, it dials a specified SIP extension. Likewise, from the Asterisk box, someone can call a SIP extension and the call goes to the intercom speaker so you can initiate a conversation with the person at the door if they just rang the bell but didn't push the intercom button. Preferably something with power over ethernet support. Thanks, -- Nick ___ --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] IAX complaints? What are they?
Griefs? rejected connect attempt from 111.111.111.111, who was trying to reach '12345678' No authority found call rejected by 111.111.111.111: No authority found But once it works it works... I have DTMF issues with sending calls from 1.2 to what I suspect is a really old 1.4 build via IAX that then hands those calls off as SIP . But I suspect it could be fixed in the SIP configuration. This is a very isolated situation. ___ --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] G729/MOH Quality
If the majority of the MoH is queues, move the location of the queue. On Nov 28, 2007 4:42 PM, Darryl Dunkin [EMAIL PROTECTED] wrote: Does anyone have any opinions on the music on hold quality over G729? The stock files seem to sound terrible over it, this is enhanced further by calls coming from the PSTN via a Zaptel gateway. I am only using the stock wav files and have not attempted to use much else so far. I've ruled out timing issues on the system generating the MOH itself (ztdummy on the PBX itself, our Zaptel gateway is a separate Asterisk server). There is no transcoding going on in the middle except via our Zaptel/T1 gateway. When using G711 it sounds fine of course, but this doesn't work well for remote sites with lower bandwidth connections. As of now, I'm torn between getting complaints from end users about the music or killing it entirely (leaving people waiting in queues with dead silence). ___ --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] Only call me once
Anyone have an idea how to implement a phone number that can only be called once? The first time it will process normally and any subsequent calls will be rejected. ___ --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] Unable to lookup host in c= line,
[Nov 28 15:42:41] WARNING[4098]: chan_sip.c:4957 process_sdp: Unable to lookup host in c= line, 'IN IP4 50045' Anyone have this problem when using T.38 faxing... and some solution perhaps? ___ --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] [Copfilter] Copy of quarantined email - *** SPAM *** [6.6/6.0] Problems getting Asterisk to detect call in
I have installed an Asterisk 1.4 on Suse93 using a FritzCard.
Some calls are logged to the ISDN log, but Asterisk is not detecting
incoming calls.
I wonder whether some other device or process is preventing Asterisk
from gaining access to the isdn line?
Is there some way to ensure that only Asterisk can listening to the
line, or get it to share the line with some other device, such as the
fax system or some other thing?
Any ideas?
Here some of the conf files.
output of capiinfo command
Number of Controllers : 1
Controller 1:
Manufacturer: AVM GmbH
CAPI Version: 2.0
Manufacturer Version: 3.11-07 (49.23)
Serial Number: 101
BChannels: 2
Global Options: 0x0039
internal controller supported
DTMF supported
Supplementary Services supported
channel allocation supported (leased lines)
B1 protocols support: 0x411f
64 kbit/s with HDLC framing
64 kbit/s bit-transparent operation
V.110 asynconous operation with start/stop byte framing
V.110 synconous operation with HDLC framing
T.30 modem for fax group 3
Modem asyncronous operation with start/stop byte framing
B2 protocols support: 0x0b1b
ISO 7776 (X.75 SLP)
Transparent
LAPD with Q.921 for D channel X.25 (SAPI 16)
T.30 for fax group 3
ISO 7776 (X.75 SLP) with V.42bis compression
V.120 asyncronous mode
V.120 bit-transparent mode
B3 protocols support: 0x80bf
Transparent
T.90NL, T.70NL, T.90
ISO 8208 (X.25 DTE-DTE)
X.25 DCE
T.30 for fax group 3
T.30 for fax group 3 with extensions
Modem
0100
0200
3900
1f010040
1b0b
bf80
0101 0002
Supplementary services support: 0x03ff
Hold / Retrieve
Terminal Portability
ECT
3PTY
Call Forwarding
Call Deflection
MCID
CCBS
==
/etc/isdn/isdn.conf
=
#SuSEconfig.isdn modified unknown
# example of /etc/isdn/isdn.conf
#
# More information: /*
[GLOBAL]
COUNTRYPREFIX = +
COUNTRYCODE = 44
AREAPREFIX = 0
AREACODE = 20
[VARIABLES]
[ISDNLOG]
LOGFILE = /var/log/isdn.log
ILABEL = %b %e %T %ICall to tei %t from %N2 on %n2
OLABEL = %b %e %T %Itei %t calling %N2 with '%n0'
REPFMTWWW = %X %D %17.17H %T %-17.17F %-20.20l SI: %S %9u %U %I %O
REPFMTSHORT = %X%D %8.8H %T %-14.14F%U%I %O
REPFMTNIO = %X %D %16.16H %T %-25.25F %U
REPFMT = %X %D %16.16H %T %-16.16F %7u %U %I %O
###
#
# You can set a daily limit for phone cost for the ISDN interface here.
# Please note following points and also read the isdnlog documentation:
#
# 1. This function may fail for many reasons, here is no guarantee that
#this protect you against high cost.
#Please be very carefully if you enable dial on demand !!!
#
# 2. Neither SuSE Linux AG nor the authors of the software are responsible
#for any damage or costs you have if you use or not use this feature.
#
# 3. If the charges are going above the limit /etc/isdn/stop is called
#and depending on the amount following actions are done:
# - 0..1 Euro above limit : short warning with 2 beeps
# - 2 Euro above limit : longer warning with 3 beeps
# - 3..4 Euro above limit : warning with 5 beeps shutdown isdn
# network interfaces
# - = 5 Euro above limit : reboot PC
#
#If you like other actions or values please modify /etc/isdn/stop
#
# 4. The number of your provider need an entry in /etc/isdn/callerid.conf,
#without CHARGEMAX has no effect.
#
# 5. Since it can cause unwanted network shutdowns or reboots, CHARGEMAX
#is disabled by default
#
###
# CHARGEMAX = 50.00
CURRENCY = 0.062,EUR
COUNTRYFILE = /usr/lib/isdn/country.dat
RATECONF= /etc/isdn/rate.conf
# replace the xx in the next 3 lines with your country's letters!
RATEFILE= /usr/lib/isdn/rate-xx.dat
HOLIDAYS= /usr/lib/isdn/holiday-xx.dat
ZONEFILE= /usr/lib/isdn/zone-xx-%s.cdb
DESTFILE= /usr/lib/isdn/dest.cdb
==
/etc/capi.conf
===
#SuSEconfig.isdn generated
# card fileproto io irq mem cardnr options
fcpci - - - - - 1
===
/etc/asterisk/capi.conf
===
;
; CAPI config
;
;
; general section
[general]
nationalprefix=0
internationalprefix=00
rxgain=1.0 ;linear receive gain (1.0 = no change)
txgain=1.0 ;linear transmit gain (1.0 = no change)
language=de ;set default language
;ulaw=yes;set this, if you live in u-law world instead of a-law
;jb. ;with Asterisk 1.4 you can configure jitterbuffer,
;see Asterisk documentation for all jb* setting available.
;mohinterpret=default ;Asterisk 1.4: default music on hold class when
placed on hold.
; interface sections ...
[asterisk-users] [Copfilter] Copy of quarantined email - *** SPAM *** [7.4/6.0] Re: Asterisk - Nortel Phone Switch
[asterisk-users] Asterisk - Nortel Phone Switch Date: Thu, 29 Nov 2007 07:52:17 + (GMT) X-Mailer: sendEmail-1.52 MIME-Version: 1.0 Content-Type: multipart/mixed; boundary=MIME delimiter for sendEmail-20854.4017086787 This is a multi-part message in MIME format. To properly display this message you need a MIME-Version 1.0 compliant Email program. --MIME delimiter for sendEmail-20854.4017086787 Content-Type: text/plain; charset=iso-8859-1 Content-Transfer-Encoding: 7bit What LAN and you using? ELAN or HSP Are you trying to connect to a signaling server? Please provide Nortel config. Jonn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, November 28, 2007 2:06 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk - Nortel Phone Switch Still trying to make my Asterisk PBK talk to our Nortel Phone Switch (C15k). Nortel did an upgrade which changed a bunch of things today, so I thought I'd give it another shot. It looks like I'm much closer this time, but still no go. Can't do calling in either direction. Anyone have any ideas? Thanks! Shawn [nortel] host=10.0.0.10 insecure=very type=peer qualify=no canreinvite=no dtmfmode=rfc2833 fromuser=user username=user secret=123 disallow=all allow=gsm allow=ulaw allow=alaw dtmfmode=rfc2833 usereqphone=yes context=from-nortel asterisk*CLI sip debug ip 10.0.0.10 SIP Debugging Enabled for IP: 10.0.0.10 The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead. Audio is at 192.168.10.2 port 17492 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.0.10:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK489024dd;rport From: Shawn Ip sip:[EMAIL PROTECTED];tag=as25dd7670 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 28 Nov 2007 18:24:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 287 v=0 o=root 3386 3386 IN IP4 192.168.10.2 s=session c=IN IP4 192.168.10.2 t=0 0 m=audio 17492 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk*CLI --- SIP read from 10.0.0.10:5060 --- SIP/2.0 486 Busy Here From: Shawn Ipsip:[EMAIL PROTECTED];tag=as25dd7670 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.10.2:5060;rport=5060;branch=z9hG4bK489024dd User-Agent: Asterisk PBX Max-Forwards: 70 Supported: replaces Date: Wed, 28 Nov 2007 18:24:14 GMT Allow: NOTIFY Content-Type: application/SDP Content-Length: 287 v=0 o=root 3386 3386 IN IP4 192.168.10.2 s=session c=IN IP4 192.168.10.2 t=0 0 m=audio 17492 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv - --- (13 headers 14 lines) --- Transmitting (no NAT) to 10.0.0.10:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK489024dd;rport From: Shawn Ip sip:[EMAIL PROTECTED];tag=as25dd7670 o: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ___ --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 . Still trying to make my Asterisk PBK talk to our Nortel Phone Switch (C15k). Nortel did an upgrade which changed a bunch of things today, so I thought I'd give it another shot. It looks like I'm much closer this time, but still no go. Can't do calling in either direction. Anyone have any ideas? Thanks! Shawn [nortel] host=10.0.0.10 insecure=very type=peer qualify=no canreinvite=no dtmfmode=rfc2833 fromuser=user username=user secret=123 disallow=all allow=gsm allow=ulaw allow=alaw dtmfmode=rfc2833 usereqphone=yes context=from-nortel asterisk*CLI sip debug ip 10.0.0.10 SIP Debugging Enabled for IP: 10.0.0.10 The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead. Audio is at 192.168.10.2 port 17492 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP
[asterisk-users] [Copfilter] Copy of quarantined email - *** SPAM *** [6.5/6.0] Asterisk - Nortel Phone Switch
Still trying to make my Asterisk PBK talk to our Nortel Phone Switch (C15k). Nortel did an upgrade which changed a bunch of things today, so I thought I'd give it another shot. It looks like I'm much closer this time, but still no go. Can't do calling in either direction. Anyone have any ideas? Thanks! Shawn [nortel] host=10.0.0.10 insecure=very type=peer qualify=no canreinvite=no dtmfmode=rfc2833 fromuser=user username=user secret=123 disallow=all allow=gsm allow=ulaw allow=alaw dtmfmode=rfc2833 usereqphone=yes context=from-nortel asterisk*CLI sip debug ip 10.0.0.10 SIP Debugging Enabled for IP: 10.0.0.10 The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead. Audio is at 192.168.10.2 port 17492 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.0.10:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK489024dd;rport From: Shawn Ip sip:[EMAIL PROTECTED];tag=as25dd7670 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 28 Nov 2007 18:24:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 287 v=0 o=root 3386 3386 IN IP4 192.168.10.2 s=session c=IN IP4 192.168.10.2 t=0 0 m=audio 17492 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk*CLI --- SIP read from 10.0.0.10:5060 --- SIP/2.0 486 Busy Here From: Shawn Ipsip:[EMAIL PROTECTED];tag=as25dd7670 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.10.2:5060;rport=5060;branch=z9hG4bK489024dd User-Agent: Asterisk PBX Max-Forwards: 70 Supported: replaces Date: Wed, 28 Nov 2007 18:24:14 GMT Allow: NOTIFY Content-Type: application/SDP Content-Length: 287 v=0 o=root 3386 3386 IN IP4 192.168.10.2 s=session c=IN IP4 192.168.10.2 t=0 0 m=audio 17492 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv - --- (13 headers 14 lines) --- Transmitting (no NAT) to 10.0.0.10:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK489024dd;rport From: Shawn Ip sip:[EMAIL PROTECTED];tag=as25dd7670 o: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ___ --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 . v=0 o=root 3386 3386 IN IP4 192.168.10.2 s=session c=IN IP4 192.168.10.2 t=0 0 m=audio 17492 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk*CLI --- SIP read from 10.0.0.10:5060 --- SIP/2.0 486 Busy Here From: Shawn Ipsip:[EMAIL PROTECTED];tag=as25dd7670 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.10.2:5060;rport=5060;branch=z9hG4bK489024dd User-Agent: Asterisk PBX Max-Forwards: 70 Supported: replaces Date: Wed, 28 Nov 2007 18:24:14 GMT Allow: NOTIFY Content-Type: application/SDP Content-Length: 287 v=0 o=root 3386 3386 IN IP4 192.168.10.2 s=session c=IN IP4 192.168.10.2 t=0 0 m=audio 17492 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv - --- (13 headers 14 lines) --- Transmitting (no NAT) to 10.0.0.10:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK489024dd;rport From: Shawn Ip sip:[EMAIL PROTECTED];tag=as25dd7670 o: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ___ --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 . v=0 o=root 3386 3386 IN IP4 192.168.10.2 s=session c=IN IP4 192.168.10.2 t=0 0 m=audio 17492 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv - --- (13 headers 14 lines) --- Transmitting (no NAT
Re: [asterisk-users] Recommendations for 100 Wifi SIP phone setup
On Nov 26, 2007 9:52 AM, Alberto Pastore [EMAIL PROTECTED] wrote: I also found the Pirelli DP-L10 dual phone to be an excellent sip client with good roaming support and discrete battery saving capability. (Used in a 14-cell wifi network with 40 cellphones). I don't know what to say I have not used the Pirelli phone but at the same time it is the same ODM as most of the Linksys and D-Link phone and I have not been too pleased with those. They work. They roam ok but they also lock up every so often and the call quality isnt the best. You can tell the G729 codec is very taxing on the device it can take 2 sec for the phone to respond to a keypress. http://www.wneweb.com/Datacom/VoWLAN.htm http://www.wneweb.com/Mobile/Dual_Net.htm RRPB-81 = Linksys WIP300 SRP8-01 = Dlink DPH-540 3Com 3C10408A etc ___ --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] Asterisk version survey
Your form can no longer accept submissions. SuSe 10.1 with latest Asterisk 1.2 using our own patches. We are about ready to go live with new installations of SLES or CentOS + Asterisk 1.4 just need to work out the bugs. On Nov 26, 2007 5:14 AM, randulo [EMAIL PROTECTED] wrote: Hi, I'd like to invite all asterisk users to answer two questions on this form: http://food4wine.ning.com/poll 1) What version do you use in production (1.2, 1.4 or both) 2) and what distro(s) It'll just take a second and the results are public and live (link on the page above) ___ --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] Recommendations for 100 Wifi SIP phone setup
Hi all, Im preparing a quote for a 5 Star hotel, planning to have around 100 SIP Wifi phones for PBX operations running on 100 AccessPoints. Network is running in ARUBA Networks - AP70 access points. The initial recommendation is to go for Hitachi Wifiphones, but i would like to know from the group the recommendations. Im planning to put up Asterisk as the PBX, Please advice me the do's and donts as i'm not experienced on such heavy installation which are mission critical. I had been using asterisk on small profiles and this would be my first Pro setup with wifi handsets if all goes as planned. the Key Questions are Is Asterisk good enough? or do we need a another Proxy like SER? What is the experience with Hitachi Wifi phone's? Any specific Issues? Any such installations done? Please do a detail Looking for experiences.. Thanks Sunil Charly Manager - Business Planning KOLTELECOM ___ --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] Recommendation for 100 SIP WiFi phone setup
Hi all, Im preparing a quote for a 5 Star hotel, planning to have around 100 SIP Wifi phones for PBX operations running on 100 AccessPoints. Network is running in ARUBA Networks - AP70 access points. The initial recommendation is to go for Hitachi Wifiphones, but i would like to know from the group the recommendations. Im planning to put up Asterisk as the PBX, Please advice me the do's and donts as i'm not experienced on such heavy installation which are mission critical. I had been using asterisk on small profiles and this would be my first Pro setup with wifi handsets if all goes as planned. the Key Questions are Is Asterisk good enough? or do we need a another Proxy like SER? What is the experience with Hitachi Wifi phone's? Any specific Issues? Any such installations done? Please do a detail Looking for experiences.. Thanks Sunil Charly Manager - Business Planning KOLTELECOM ___ --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] Digium and Asterisk
Actually if you rule out all the clone tormenta cards (nothing wrong.. but very dated design... I wouldnt buy one today) the Digium cards aren't too expensive. Those tormenta cards are the ones you see for $300-400 typically. Some people like Digium others Sangoma. Personally I'm a Sangoma man. Some people report certain main boards and Dell servers aren't compatible with some digium cards. According to a post here on the mailing list someone from Digium implied that they will replace cards with these conflicts with newer model card that does not have these conflicts... your millage may vary I don't believe that forum posting was made in any official capacity but I also doubt that Digium would not do something to correct an issue for an item under warranty. On Nov 22, 2007 8:03 AM, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; Is Digium the best telephony cards to be used with Asterisk? The prices are some how high, any suggestion? Regards Bilal Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ --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] Vicidial + Unicall mfcr2
Dear Bruno, I had the experience of using the Vcidial with the boards of Digivoice. It worked very well! Leonardo Silva Does Vicidial work together with Unicall/mfcr2 ? Best Regards -- Bruno de Assumpção Loureiro msn: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Aastra 480i CT - No Incoming Calls
I just bought an Aastra 480i CT for a client who needed cordless capabilities in their office. I'm trying to set up the base station and cordless handset in my office first. I'm able to connect the phone to my Asterisk box and make outgoing calls from either the base station or the handset - to extensions within my office as well as numbers outside the network. But I can't receive calls on either the base station or the handset. All of the calls go strait to voice mail. I've never had this problem with the phones I use in my office - Linksys SPA942. What am I doing wrong? Thanks, Danny ___ --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] Aastra 480i CT - No Incoming Calls
I figured it out. Unlike the Linksys SPA942, the Web GUI interface for configuring the phone requires Proxy Server as well as the Registrar Server fields be populated with the IP address of the Asterisk server. [EMAIL PROTECTED] wrote: I just bought an Aastra 480i CT for a client who needed cordless capabilities in their office. I'm trying to set up the base station and cordless handset in my office first. I'm able to connect the phone to my Asterisk box and make outgoing calls from either the base station or the handset - to extensions within my office as well as numbers outside the network. But I can't receive calls on either the base station or the handset. All of the calls go strait to voice mail. I've never had this problem with the phones I use in my office - Linksys SPA942. What am I doing wrong? Thanks, Danny ___ --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] spandsp as T.38 termination?
You need a T38 gateway of sorts, sort of like the app_t38gateway of CallWeaver. However digium refuses to include such a program with Asterisk. On Nov 21, 2007 6:13 PM, Robert Moskowitz [EMAIL PROTECTED] wrote: It seems that Spandsp has everything in it (when you include rxfax and txfax) to be a T.38 termination when used with Asterisk 1.4? And if so, what version of Spandsp? What version of IAXModem (so I don't have to also deal with T38Modem)? ___ --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] Help: How to configure SIP domain on SPA942
Take a look at the admin guides at http://spc.pifiu.com On Nov 18, 2007 10:53 PM, Philip Prindeville [EMAIL PROTECTED] wrote: I'm using a bunch of SPA942's, and I'm trying to provision them mostly by DHCP (and what I can't set that way, I try to provision via HTTP interface into the phone). I changed the domain in my AstLinux config from astlinux to redfish-solutions.com, and set that in my sip.conf file as well: context=incoming canreinvite=no realm=redfish-solutions.com domain=redfish-solutions.com,incoming-redfish tos=184 disallow=all allow=ulaw allow=gsm localnet=192.168.10.0/255.255.255.0 externip=X.X.X.X (Footnote: do I need a default context? I'd rather not having one... I'd rather specify where my calls go explicitly...) However, my phones don't seem to be registering with any (symbolic) domain... just the IP address of their DHCP or TFTP server (can't tell which, since it's the same box). -- SIP read from 192.168.10.187:5060: REGISTER sip:192.168.10.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.187:5060;branch=z9hG4bK-1e31e66f From: sip:[EMAIL PROTECTED];tag=e798d04e1a8af3a6o0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 58671 REGISTER Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:5060;expires=3600 User-Agent: Linksys/SPA942-5.1.15(a) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces pbx2*CLI --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.10.187 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.10.187:5060: SIP/2.0 404 Not found (unknown domain) Via: SIP/2.0/UDP 192.168.10.187:5060;branch=z9hG4bK-1e31e66f;received=192.168.10.187 From: sip:[EMAIL PROTECTED];tag=e798d04e1a8af3a6o0 To: sip:[EMAIL PROTECTED];tag=as7c1c3fa2 Call-ID: [EMAIL PROTECTED] CSeq: 58671 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 The config seems to take: Our local SIP domains: Context Set by redfish-solutions.comincoming-redfish [Configured] So, what's the DHCP option (or the HTTP knob) to tweak to get the phones to think they are in the redfish-solutions.com domain? Thanks, -Philip ___ --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] function voicemailmain
You need some experiance with the ANSI C programming language. Once you have acquired that the rest is pretty straightforward. On Nov 14, 2007 2:21 AM, Rilawich Ango [EMAIL PROTECTED] wrote: You mean modify the source? Could you give me an example, say I wrong to remove advance option? On Nov 14, 2007 1:59 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: vi app_voicemail.c On Nov 13, 2007 10:34 PM, Rilawich Ango [EMAIL PROTECTED] wrote: Hi all, Can I simply the voicemailmain IVR? I just only want some of the option in voicemailmain, ie read or delete messages. Is it possible to configure that function? Ango ___ --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 ___ --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] Linksys 942 Call Transfer
did you try canreinvite=no in your sip.conf file It would also help to: 1) Post the relevant configuration files (phone AND Asterisk) 2) Post the EXACT message from column 1 to EOL 3) What version of Asterisk? Stock? From a certain distribution? Patches? Or I could just say There is a problem with your configuration, transfer of calls from an SPA-phone works fine for me. (it really does!) ___ --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] function voicemailmain
vi app_voicemail.c On Nov 13, 2007 10:34 PM, Rilawich Ango [EMAIL PROTECTED] wrote: Hi all, Can I simply the voicemailmain IVR? I just only want some of the option in voicemailmain, ie read or delete messages. Is it possible to configure that function? Ango ___ --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] 'Traditional' Faxing
As another suggested the Sangoma cards should work. However we need someone to write a frontend to Steve Underwood's wonderful spanDSP library. This will allow us a T38 gateway of sorts meaning you can connect a Linksys ATA using T.38 and we can say that (assuming your fax machine strictly complies with the relevant standards) faxing will work with 100% reliability, thats a bit more assuring that it should work, no? On Nov 10, 2007 7:34 AM, Greg Cockburn [EMAIL PROTECTED] wrote: Hi all, the company I work for has an aging Digital PBX attached to an E1. This PBX has a few analogue lines, one of which we use a 'traditional' fax machine on. I want to upgrade our PBX and Asterisk is almost a perfect fit. The only problem I can't seem to find a working solution for is Faxing. I don't want to use Hylafax or other similar methodologies. I believe there maybe someway to bridge an Analogue FXS port to a channel on the E1? Basically I want to mimic what we have now. 1. Any person can send a fax using the fax machine, and the PBX picks the next free channel on the E1. 2. A fax call can come over any channel on the E1, and the dialed number is matched and sent to the analogue FXS port of the PBX to be received by the fax machine. Is there anyway I can do this in Asterisk that will work seamlessly? I have not yet purchased any hardware, so recommendations would be greatly appreciated. (I believe some of the problem exists due to timing, does any hardware; E1 card / Analogue card; support linking a timing signal together?) Sangoma, Digium, Pika? Thanks all for any help on this one. Greg. ___ --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] Kernel Native PCIE Network Cards?
Broadcom BCM5721 working here with SuSE (2.6.16) On Nov 9, 2007 12:39 AM, Michael J. Liberatore [EMAIL PROTECTED] wrote: Hi, I am getting a new sangoma t1 card soon and that will max out my slots, which means i need to take out a card. I am going to take out my pci network interface card (10/100) I have an open pci-e slot i have never used in the machine so i am going to buy a pci-e 10/100 or gigabit network adapter. I want to find one that works natively with the linux kernel. I hate using hardware that requires additional drivers in linux and have read tons of nightmares of people trying to get pci-express nic drivers to work with linux. So if someone could point me to a card that is natively supported in 2.6.15 i would appreciate it. Thanks Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] Wanted: tutorial on troubleshooting SIP issues
On Nov 8, 2007 7:11 PM, Philip Prindeville [EMAIL PROTECTED] wrote: Anyway, pointers for someone wanting to learn to quickly diagnose SIP conversations would be great. Read rfc2543, rfc3261 rfc3265. Otherwise what you want to do is akin to trying to diagnose a nuclear reactor and not wanting to learn about nuclear physics! ___ --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] 7960 Queue Issue
Setup a 2nd registration on the phone that only allows 1 call at a time. Ideal setup it up as a shared appearance so call forwarding, etc dont work on that registration. This way your phone has 2 registrations 1 for any direct call and another for shared calls, queues, etc. ___ --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] Asterisk 1.4 + Presence
* hint: The 'hint' priority associates an extension with an Asterisk channel for the purpose of mapping the state of the channel to a state of the extension. In asterisk, a channel (technology/device) can have several states (unavailable, in-use, busy, ringing, etc) but an extension is just a label for a sequence of applications. However, when communicating the state of the channel to an external device, such as a receptionist console, you cannot use the Asterisk internal channel names, but must use an externally identifiable resource name, typically the extension number. A device would then subscribe to the state of the extension of interest and receive status notifications from the supporting technology channel. This is used in the SIP channel (implemented via the SUBSCRIBE/NOTIFY mechanism of RFC-3265) to light up the status lamps on SIP phones. This is supported in SNOM phones (see also) with their programmable keys set to type destination, as well as in Polycom (500/600), Aastra ( 480i, 9133i ), and Sayson phones. It is also supported in Citel SIP Handset Gateways. Privacy considerations: In sip.conf you can define a subscribecontext= value that determines in which context Asterisk should search for the matching extension when a subscribe request is received from the phone; however, if the extension doesn't exist in that context Asterisk is going to look for it in the default context! In other words: Everyone can subscribe to a hinted extension that is defined in the default context. By the way, specifying an empty subscribecontext is also fine if the phone should not at all subscribe to _any_ context. Likewise bug/patch 5515 (post Asterisk 1.2.0) adds devstate support also for MGCP (so far SIP, IAX and ZAP are supported; show channeltypes tell you which channels in your Asterisk support device status notification). Question: Does this patch only show a device which is unavailable (e.g. disconnected), or does it also show busy? Answer: Also busy (in use). Also chan_capi-cm v0.6.2 and later comes with basic hint support. It appears, however, that the dynamic naming of CAPI channels that includes the called number makes monitoring of a CAPI line for outgoing calls practically impossible - at least for now. Note: the 3rd party Bristuff patches come with app_devstate that permits state manipulation through the dialplan. New: While Asterisk 1.6 will include func_devstate natively there is now also a backport available for 1.4. This is quite similar to app_devstate as part of the bristuff patches. Example exten = 200,hint,SIP/phone1 ; this is case sensitive (!) in 1.0.9 and 1.2.0 exten = 200,1,Macro(stdexten,SIP/phone1) If you want to monitor the state of multiple phones using one speeddial, you can do so: exten = 200,hint,SIP/201SIP/202SIP/203 Asterisk seems to provide syntax for allowing more than one channel to be mapped to any particular extension with the hint system. Useful CLI commands for debugging are SIP show subscriptions, show hints, show channeltypes and SIP show inuse. On Nov 6, 2007 1:36 PM, Alejandro Cabrera Obed [EMAIL PROTECTED] wrote: Dear all, I'm using Asterisk 1.4 as my SIP server of my LAN users. The SIP clients are using different operating systems such Debian, Gentoo and Windows XP so they use different SIP softphones like SJPhone, Twinkle and X-Lite. In order to let SIP clients to see the presence status to each other, do I have to establish any special setting in Asterisk 1.4 ??? Or the presence status (online, offline, away, etc.) is only up to the SIP clients and not up to the Asterisk ??? Really thanks Alejandro ___ --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] Asterisk Help
On Nov 6, 2007 2:25 PM, Jarga Jallow [EMAIL PROTECTED] wrote: Under asterisk info: Sip registry 12/12 76.xxx.xxx.xxx D N 5066 UNREACHABLE 11/11 76.xxx.xxx.xxx D N 5064 UNREACHABLE 10/10 76.xxx.xxx.xxx D N 5062 UNREACHABLE All these IP phones are behind NAT. What could be the problem? You aren't supposed to be registering to your IP phones you should have the IP phones registering against your 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] Wifi handover/roaming
For fastest handover disable any sort of encryption and use the same SSID for all AP... infact I don't know how you would setup roaming otherwise. Channels don't have to be the same, but optimize for the best RF performance/least channel overlap. ___ --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] OT: Aastra 57i configuration via TFTP problem
Have you looked at your TFTP server logs? On 11/7/07, Roi Stork [EMAIL PROTECTED] wrote: I am currently testing a 57i unit. No problems configuring the phone's config via phone/web UI. We are trying to avoid using the web UI, the reason is it will take a long time typing the softkey xml applications URIs on each phone, so we chose TFTP. Tried configuring the phone via a TFTP config server, but no changes took effect. I wonder why it doesn't work with TFTP even if I was able to upgrade the firmware via the same method. Here's how I set it up, maybe someone can point where I did it wrong: 1) No DHCP, so I manually set the network settings via phone UI. 2) The files aastra.cfg and mac address.cfg are in the TFTP root folder. 3) Restarted the phone. ___ --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] grandstream troubles
Have you tried a second unit? I don't trust the Grandstream ATA at all. We only bought 3 but none worked! On 11/7/07, Per Jessen [EMAIL PROTECTED] wrote: I've got a Grandstream 487 in a home-office. The phone-side is working fine, but the user is complaining that his internet connection keeps disappearing. The Grandstream is set up as NAT router, and there's just one PC hanging off the LAN. Has anyone experienced anything similar? /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --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] Linksys SPA-941 Unavailable
Add qualify=5000 in the relevant section of your sip.conf (under the [6464]) and also make sure the phone is configured NAT Keep Alive Enable = YES. On 11/7/07, Kim Joung-il [EMAIL PROTECTED] wrote: Sorry guys, I should have already sent such details...so 1. Yes, device is behind NAT (for ram) 2. Bellow is sip.configuration file [general] bindport=5060 bindaddr=0.0.0.0 context=invalid-context musicclass=default externip=56.236.64.79 allowguest=no useragent=PBX maxexpirey=7200 defaultexpirey=3600 realm=PBX progressinband=never disallow=all allow=ulaw allow=alaw register = pbx1:[EMAIL PROTECTED] [6464] type=friend dtmfmode=rfc2833 context=default nat=yes canreinvite=no qualify=2000 host=dynamic callgroup=7 pickupgroup=7 username=6464 secret=6464 disallow=all allow=ulaw allow=alaw allow=g729 callerid=Frist Name 6464 subscribecontext=hints [EMAIL PROTECTED] accountcode=6464 [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Post the relevant configuration files we'd be glad to help. On 11/6/07, Kim Joung-il wrote: Hello! We are using several Linksys SPA-941 in our office. After IP change occur devices seems not to be reachable, actually unavailable! Devices is connected, e.g. we can place a call using SPA-941 but can not receive any calls... Kim __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.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
Re: [asterisk-users] Video Call
It should be possible to get the video call over PRI or ISDN and depending on the codec in theory it could just be throwing packets into SIP. On 11/1/07, voip Server asterisk [EMAIL PROTECTED] wrote: Hi.. Iam new with asterisk PBX, and i have read about asterisk video call.: my question: 1. Is imposible to establish system video call (from Phone with GPRS/3G enabled to Computer Running Softphone like X-Lite) over Asterisk Gateway.. 2. If posible what requirement (Hardware and Software on my Asterisk,PC or My Phone) Thanks Joko Pitoyo ___ --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] Determination of billsec
That's odd because in my world I *NEVER* have a CDR show ANSWERD and anything besides 1 billing seconds. Also -- Dave shows up with the stuff and isn't confused about his name. CSB -- I'd say the reason you are having this problem is you are dialing a local channel. Have you tried otherwise? Which version of Asterisk? On 11/7/07, Doug [EMAIL PROTECTED] wrote: At 02:47 11/7/2007, CSB wrote: Content-Type: multipart/alternative; boundary==_NextPart_000_0007_01C82187.BC96F350 Content-Language: en-nz How is the billsec field calculated in CDRs? I have a situation where billsec is being reported as 0 despite the call being answered and a conversation occurring. An example record follows: '2007-11-06 21:36:50', '6495566778', '6495566778', '0116495566778', '1100012_1', 'Local/[EMAIL PROTECTED],2', 'SIP/64.192.001.001-08893238', 'Dial', 'SIP/[EMAIL PROTECTED]||hH', 10, 0, 'ANSWERED', 3, '', '1194338210.61', '' Where did you get this CDR? CDRs should look more like: http://www.asterisk.org/doxygen/1.2/AstCDR.html clidCaller ID src Source dst Destination dcontextDestination context channel Channel name dstchannel Destination channel lastapp Last app executed lastdataLast app's arguments start Time the call started. answer Time the call was answered. end Time the call ended. durationDuration of the call. billsec Duration of the call once it was answered. disposition ANSWERED, NO ANSWER, BUSY amaflagsDOCUMENTATION, BILL, IGNORE etc accountcode The channel's account code. uniqueidThe channel's unique id. userfield The channels uses specified field. A call can ring for 10 seconds, then be answered and hung up on (or dropped for some reason), and end up having billable seconds of zero. Where in this CDR is there evidence of a conversation having taken place? A conversation would at least be 15-30 seconds: Hello? Yeah, It's me--Dave. I got the stuff, man. Dave? Yeah. 'Dave'. It's me. Dave's not here. Click. ___ --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] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking
On 11/7/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: I'm not really sure if Callweaver has this limitation or not. But they did aim at using high-resolution timers from the Linux kernel. Callweaver does. Asterisk does not. I'm awaiting their next release its supposed to have proper faxing support. Besides that its just a fork of Asterisk 1.2. ___ --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] wifi
I am very happy with the Linksys WRT54GS v4 routers and the WRT54GL (which are both supposed to be the same hardware). Also the Buffalo WHR-G54S, WHR-G125 WHR-HP-G54S models all running the DD-WRT firmware of Sebastian Gottschall. However the management featureset is still that of a consumer router. However I have yet to find a WiFi handset I am happy with. On 11/6/07, Michael Graves [EMAIL PROTECTED] wrote: I'd like to survey those on-list who actually use wifi SIP handsets. What type of wifi access point do you use? Are you happy with 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] Linksys SPA-941 Unavailable
Post the relevant configuration files we'd be glad to help. On 11/6/07, Kim Joung-il [EMAIL PROTECTED] wrote: Hello! We are using several Linksys SPA-941 in our office. After IP change occur devices seems not to be reachable, actually unavailable! Devices is connected, e.g. we can place a call using SPA-941 but can not receive any calls... Kim __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.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
Re: [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking
On 11/6/07, Hans Feringa [EMAIL PROTECTED] wrote: I understood that a timing device (ztdummy if no zaptel hardware is present) was not necessary anymore with linux kernel 2.6. When I enable iax2 trunking I get this warning chan_iax2.c:8908 build_user: Unable to support trunking on user 'xx' without zaptel timing The linux kernel is 2.6.22-14-386 Can I ignore this message, and is trunking working despite this warning? The ztdummy module is not part of the zaptel ubuntu package, so it cannot be loaded. I wanted to install from ubuntu packages for a change and not compile it from source. rgds, I believe that's OpenPBX that tries to derive its timing without Zaptel devices, however then you need to recompile your Kernel with 1000Hz timing as most use ~250Hz by default. Linux 2.6 + Ztdummy works fine and I'll take that over having to recompile the Kernel any day. ___ --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] Sangoma S200 and Digium TDM400P together
What's the result if you do cat /dev/zap ? On 11/6/07, Paulo Garcia [EMAIL PROTECTED] wrote: Hi, I have these two cards, the Sangoma has 4 fxo interfaces and the digium has 1 fxo and 1 fxs. After install the sangoma card, my zaptel.conf was configured for that card. I'm trying to configure the Digium one together thinking that the Digium ports should be 5 and 8 but it doesn't works. Someone has some example about this? Thanks in advance Pauçp ___ --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] Sangoma S200 and Digium TDM400P together
Sorry I mean ls /dev/zap On 11/6/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: What's the result if you do cat /dev/zap ? On 11/6/07, Paulo Garcia [EMAIL PROTECTED] wrote: Hi, I have these two cards, the Sangoma has 4 fxo interfaces and the digium has 1 fxo and 1 fxs. After install the sangoma card, my zaptel.conf was configured for that card. I'm trying to configure the Digium one together thinking that the Digium ports should be 5 and 8 but it doesn't works. Someone has some example about this? Thanks in advance Pauçp ___ --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] MeetMe CPU resources
Just remember if you don't have any Zaptel cards you are going to have to use ztdummy to run app_meetme. Ztdummy essentially requires Linux 2.6, which you should be using anyways. On 11/6/07, Carles Pina i Estany [EMAIL PROTECTED] wrote: Hello, First of all: also thanks to Doug Lytle and Steve Edwards. Just answering one time to all of you. I had the feeling that this computer, for 15 Meetme users, was more than enough... but we wanted to avoid any last-minute surprises! Now we are more sure that everything will work fine. Ah yes, we will use VoIP, without transcoding (I hope!), without Digium Timer card (but I will check, just in case we need it) On Nov/06/2007, Tony Mountifield wrote: In article [EMAIL PROTECTED], Carles Pina i Estany [EMAIL PROTECTED] wrote: It will depend on whether you are using VoIP or a PRI card. I have a number of systems that have a single Pentium 4 @ 2.8GHz (with HT), 1GB RAM and a 4xE1 PRI card (TE410P), and they regularly have conferences with up to 90 participants. I would expect them easily to handle the full Wow, 90 participants. Do you use just MeetMe in Asterisk? Just for curiosity: All of them can talk to conference? or only some of them? I thought about it, and for me, 90 open microphone participants looks like some white noise :-) Not tried here... just wondering how do you do. Thanks! -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --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] Fax Problems with SpanDSP
On 8/29/07, Steve Underwood [EMAIL PROTECTED] wrote: Carlos Chavez wrote: On Wed, 2007-08-29 at 00:03 +0300, Tzafrir Cohen wrote: On Tue, Aug 28, 2007 at 10:11:03PM +0200, Christian Peter wrote: Hi list, I'm running current SpanDSP http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre6.tgz with Asterisk 1.2.22 somewhat successfully. Shouldn't you have used spandsp 0.0.3 with asterisk 1.2 ? Actually, as Steve Underwood has gently reminded the list several times, he recommends SpanDsp 0.0.2 for Asterisk 1.2 Well, its not so much that I recommend it. Its just that I have never done anything to adapt the app_rxfax.c and app_txfax.c for Asterisk to work with newer versions of spandsp. Compared to the current spandsp, the softFAX in 0.0.2 actually sucks. Steve I've been using SpanDSP 0.0.3 and the app_rxfax of January 2006 (I notice there is a newer one) and it works great. No problem receiving 100+ page faxes via PRI. ___ --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 over T1 calls dropping, cause 100
On 10/31/07, Michelle Dupuis [EMAIL PROTECTED] wrote: The T1 was setup as tie line, not a trunk. The Bell guy tried setting up the line 2 ways: 1. As a trunk. This did not work because: a) When he typed in the access code for the trunk on a phone set (and then any numbers), the call never appeared on the Asterisk side. b) The Bell guy said that unless Asterisk was generating a dialtone, a trunk would not work (I struggled to understand these explanations...but figured I must be missing something) There is no dialtone on a PRI/T1. I think what he meant was you need to change in your zaptel config pri_cpe to be pri_net then it will allow you to setup that trunk. ___ --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] Mobile phone codecs ...
On 10/31/07, Gordon Henderson [EMAIL PROTECTED] wrote: Not strictly asterisk related, however... Here's an odd one for you.. I got a Nokia E90 and setup it's SIP client which runs via Wi-Fi (anyone know if I can make it work via GPRS/3G?) Anyway, in a fit of idleness, I thought I'd see what codecs it supports, as I couldn't find it in the manual... And it supports: ilbc g729 ulaw/alaw No GSM! How odd is that, given that it's a GSM mobile phone... Anyway, my quest for the ultimate one handset solution is getting closer. If Wi-Fi weren't so rubbish and my house not made of Dartmoor Granite it might have half a chance of working outside the room with the access point, however ... Anyone tried the Plantronics Voyager 510 bluetooth headsets which regsiters to both a mobile phone and their own base unit (which presumably has a USB sound device) as in: https://www.ukheadsets.co.uk/thc-plantronics-voyager-510-usb-dongle-rn-422-action-show_detail-show_products_mode-cat_click I'm not a fan of soft-phones, and not sure I want to have a borg implant on when I'm not driving, but ... Oh well... Back to the grind! Gordon ___ I think that's pointless. Why do you need a USB audio device? You can pair it to the computer directly and use it with any soft phone. ___ --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] Faxing with Asterisk SpanDSP [Was Fax Problems with SpanDSP]
Steve Underwood wrote: SpanDSP cannot be used by the standard distribution of Asterisk, as it is GPL code. However, if you are using Asterisk within the restrictions of the GPL you can add app_rxfax and app_txfax to Asterisk quite easily. I was wondering how someone could modify Asterisk to be GPL compliant? ___ --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] (no subject)
Honestly, Its my opinion that the Aastra phones are very lacking in the firmware department. If they could get that sorted out I wouldn't mind using them. But for now there are too many NAT issues mostly caused because they use an OLD version of Broadcom CallCtrl. Why they use an ancient version is beyond me but the phones dont even have a NAT keepalive option. They promise updates to their firmware but then they only fix minor bugs. Grandstream are ok. But as others have said their support is very lacking. I've had products of theirs behave very oddly like operate and refuse to apply any settings no matter what and not allow a factory reset... paperweight. I'd personally use Polycom in the situations where there's no NAT and the Linksys SPA-phones where you do have NAT. On 10/29/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) 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 ___ --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] T.38 Faxing and Asterisk
I thought there was some talk of getting T38Gateway into asterisk_addons? Stupid linking bullshits. On 10/31/07, Paul Bryson [EMAIL PROTECTED] wrote: Nasir Iqbal wrote: Hi, Have you tried Callweaver http://www.callweaver.org I was really hoping to be able to use Trixbox to do this and it's a pretty complete solution by itself. Unfortunately that requires Asterisk. It appears that there is no way to get Asterisk, or anything on the Asterisk box, to act as a T.38 endpoint. This appears to be the result of a licensing issue with SpanDSP. http://www.voip-info.org/wiki/view/T.38 That's a real shame as T.38 termination support is one of the last big pieces for us to make Asterisk a seamless solution. Paul Bryson ___ --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] G.729 required for IP---TDM---IP
Here's a link to the free version: http://asterisk.hosting.lv/ On 10/31/07, Gordon Henderson [EMAIL PROTECTED] wrote: On Tue, 30 Oct 2007, satish patel wrote: Dear all I have already post this question but i need more input for this setup [IPphone]--[Asterisk]E1---[Avaya]---[ip_Extention] Asterisk - codec (G.711/ulaw) Avaya - codec ( G.711/ulaw) Now I need G.729 on my asterisk side and i have put G.729 codec setting on my IP phone and when i make call from asterisk to Avaya Extention i got error translator not in path so i need to get license of g.729 on asterisk for transcoder or it will work wothout translator ??? My question is :-- Is there Required G.729 (License) on Asterisk Or Not ??? You can purchase them from Digium: http://store.digium.com/productview.php?category_id=5product_code=8G729CODECmain_category_id=5 $10 each. Install one license for each simultaneous g792 call you expect to take on the asterisk box and off you go. There are free versions of g729 avalable, but if your country is compatable with the various (US) patent laws then you ought to pay the license fee to stay legal. 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 ___ --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] Large voicemail
Probably the best option is store the messages in IMAP and the userdate in a database. Honestly I dont think there is an issue with any number of mailboxes the issue is going to be how many calls at once your system can handle or how well your architecture scales to handle multiple machines. Can your storage handle 5,000 mails being recorded at once? Just trying to sort out the thousand different aspects of it all in my mind right now I say you give it a try but expect to write your own voicemail fron the ground up and not necessarily based on Asterisk. Then again, I could be wrong. On 10/25/07, Pepo [EMAIL PROTECTED] wrote: I am trying to use Asterisk as the voicemail system of the TELCO where I work. I wanna test with 2 mail boxes ( and later with a better machine/server I hope try with 7 ). How do I include in voicemail.conf the file with the mail boxes?, In a big system like this,is better use text files or any database? Thanks -- Linux User Registered #232544 Jabber : [EMAIL PROTECTED] Ekiga : [EMAIL PROTECTED] ICQ : 337889406 GnuPG-key : www.keyserver.net --- dum loquimur, fugerit invida aetas: carpe diem, quam minimum credula postero. ___ --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] (no subject)
Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) 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] SIP phone recommendation (used to be: no subject)
My apologies to the list for not having entered a subject line in the email. Thanks On Oct 29, 2007, at 1:42 PM, [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) 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 ___ --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] SIP phone recommendation (used to be: no subject)
Well, just general office use. They are a real-state construction company, so the phones will get some heavy use since most of the phones are going to sales associates. Now, one of the things we are most interested in are: 1) Asterisk compatibility 2) Mass provisioning 3) Remote management 4) Excellent audio quality (I know there are many factors involved, but would like to rule out the phone set itself) 5) Robustness 6) Vendor reputation and warranties We have used Linksys 941s in the past and think they're pretty good. However, we've only used them in 3-5 phones office environments. We've also used the Polycoms IP 501 and 650s. They seem good, but sometimes the users complain about the audio being a bit weird in the sense that, probably, the silence detection may give the user a feeling that the line dropped. Then again, we've only used these once (one client installation for each), so for practical purposes, we don't really have any larger quantity real-life experience. Thanks On Oct 29, 2007, at 2:18 PM, Eric Chamberlain wrote: What is the use case? Linksys, Polycom, Snom, and Aastra all have their strengths and weaknesses. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, October 29, 2007 10:42 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] (no subject) Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) 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 ___ --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] Everyone is busy/congested: IP Trunk
No: register = abc:[EMAIL PROTECTED] [peer] host=zzz Its possible to make mistakes and typos you know. Maybe you can post your config file and we can help you. On 10/26/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi Pablo; How the IP address will be wrong, and asterisk able to do registeration on the destination? If the IP address wrong, so I will not be able to register on that IP address. Regards Bilal Hi List; Ip address to destination? Unable to create channel of type SIP (cause 3 - No route to destination) i think you have the wrong ip information I established an SIP IP Trunk between Asterisk and another softswitch (asterisk registered on the softswitch successfully) and I saw this on the softswitch. From firefly softphone, I was need to do a call to be via this softswitch (ofcourse, the softphone will send for asterisk and asterisk should route to the softswitch based on the extensions.conf configurations. But, always I receive this message (and the call does not even reach to the softswitch, it is not sended from Asterisk to the softswitch): Executing [EMAIL PROTECTED]:1] Dial(SIP/EgyptOeratorSIP-09f9bed0, SIP/[EMAIL PROTECTED]) is new stack Unable to create channel of type SIP (cause 3 - No route to destination) Everyone is busy/congested at this time (1:0/0/1) Anyone faced that? Is it related to a paramater that control number of allowed channels per IP trunk? Maybe I have such parameters is 0 ? I do not know even if there is such parameter. At the softswitch, I do not see even any attempt (nothing related to the dialed number), so why Asterisk does not send the called number to the softswitch and why asterisk assume there is not available channel? The softphone codec is g729a and the softswitch support such codec. Also, if it is a codec matter, then call should be send to the softswitch, and the softswitch will gives an error related to the codec missmatch. Any help? Regards Bilal Ghayad __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.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
Re: [asterisk-users] XML file for spa devices
Take a look at http://spc.pifiu.com there they have the spc.exe ( Linux variant) which will generate the sample XML file for your firmware version. There is also in PDF format the admin guides that explain all the parameters. On 10/29/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, i need an XML file format which is used in remote provisioning of different spa devices. Please somebody tell me the format or tell me where can i find it on the internet. I also need a list of parameters which are configured using auto-provisioning. -- 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 ___ --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] XML file for spa devices
Or you can download them at http://spc.pifiu.com and not have to go through that bullshit. On 10/29/07, John Mason Jr [EMAIL PROTECTED] wrote: If you go to linksys's website and click on partners then apply for partnership you will be able to get access to the documents programs you need John [EMAIL PROTECTED] wrote: Take a look at http://spc.pifiu.com there they have the spc.exe ( Linux variant) which will generate the sample XML file for your firmware version. There is also in PDF format the admin guides that explain all the parameters. On 10/29/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, i need an XML file format which is used in remote provisioning of different spa devices. Please somebody tell me the format or tell me where can i find it on the internet. I also need a list of parameters which are configured using auto-provisioning. -- 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 ___ --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] SER / Asterisk and mediapath
Hi, I'm trying to have a SER machine send calls to an Asterisk server working as an IVR. I was able to do this part just fine. Also, when the caller makes certain options in the IVR, the call is then transferred to an extension via SER. This part is also just fine. However, I'm trying to get Asterisk out of the media path once the caller has made a selection in the IVR. Can anyone give me any hints? I wasn't sure if using canreinvite since I wasn't sure if that would affect the caller's interaction in the IVR. 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
