[Asterisk-Users] (OT:) Tool for trying/troubleshooting WAN/LAN
Hi folks! This is sort of OT but I thought maybe someone had a tip for me. What I'm looking for is a tool that I can install on two computer for example, put one on each side of the customers WAN and try the connection - simulalate x calls (using codec xxx) and get statistics out of it (delays, jitter, dropped packets and that sort of things). I have looked at ethertap (to debug the LAN traffic) and iperf but I don't know if it might help me. Is there anyone that have done such things and have any tip on software to use? Thanks a lot for any reply! Regards, Lars Fredriksson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting User Info in extensions.conf? (ZyXEL P2000W)
Hi! I'm having a ZyXEL P2000W that I'm using together with my Asterisk box (CVS from some week ago). When I get a call directly to the *-box ([EMAIL PROTECTED]) I see on the console that Asterisk get the calling users name as CIDName and his SIP-address as CIDNum - but at the P2000W i only get asterisk as CLIP? And I can see in the P2000W that asterisk is received as User Info. Is it possible to set the User Info from extensions.conf or is there any other solution for this? Any thoughts? Lars --- http://www.fredriksson.net/ mailto:lars at fredriksson dot net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queues - announcements and not busy members
Kevin P. Fleming skrev: Lars Fredriksson wrote: I have benn playing a little with quesues tonight and I found out if there are at least one member-extension free the announcement with p'the place in the queue wont be played to the person who called in. This is a change that went into CVS (and changed the default behavior, which I disagree with). If you are running CVS HEAD, there is a makefirstannouncement variable in the loop that handles the first caller, and it defaults to off. I don't know if the person who added this added a config option to control it, but I don't believe they did. In any case, you can look in the code and change the initial value of that variable to 1, which will return the behavior to the way it used to be. Great! Thanks for the hint! Lars --- http://www.fredriksson.net/ mailto:lars at fredriksson dot net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming calls from I-net only for IP-address?
Hi! I'm trying to set up the possibility for users to call my Asterisk from the net. The Asterisk is behind a Intertex IX66 in which a have set Static domains so it forward all calls for my hostname and external IP to the * box. when somenone calls at lars@external-ip it all works, but if they call lars@hostnmame it wont work. At my Asterisk console I get the following output; --SNIP-- Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.5.253:5060;branch=z9hG4bKb6576efa3370cb80a7894da3a985956f.0 Via: SIP/2.0/UDP callers IP:5060;branch=z9hG4bK1B65B3770FC245EAA09D24BE6574A345;received=callers IP;rport=5060;branch=z9hG4bK1B65B3770FC245EAA09D24BE6574A345 From: Name sip:username@bbtele.se;tag=2152369980 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 11968 ACK Max-Forwards: 69 Content-Length: 0 10 headers, 0 lines Destroying call '[EMAIL PROTECTED]' --SNAP Any thoughts or anyone having the same problem? I can't figure out why it wont work! Regards, Lars --- http://www.fredriksson.net/ mailto:lars at fredriksson dot net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] queues - announcements and not busy members
Hi! I have benn playing a little with quesues tonight and I found out if there are at least one member-extension free the announcement with p'the place in the queue wont be played to the person who called in. Is this possible to change so the announcement will be played even if there are free member-extensions? I think that would be nice (well it's not how ACD-groups usually works but anyway) Best regards, Lars --- http://www.fredriksson.net/ mailto:lars at fredriksson dot net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 vs 7905
Hi! I'm also going to buy an Cisco phone for testing with * - and my question is which one? For the 7960 for example I have the following packages to select in between: Cisco IP Phone 7960G, Global, Spare Cisco IP Phone 7960G Global 7960 IP Phone With One CallManager Express Station 7960 IP Phone with one Station User License The same packages are available for the other models also. Is there anyone that can explain the differences between the packages to me, I think the best would be to get an with a SIP image directly (as I've understood that it wotks best with *) but that isn't possible maybe? Thanks for any advice! Best regards, Lars Fredriksson --- Lars Fredriksson Ockelbo, Sweden mailto:[EMAIL PROTECTED] http://www.fredriksson.net/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of David Croft Sent: Monday, March 22, 2004 9:16 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 vs 7905 If you look at http://www.cisco.com/image/jpeg/en/us/guest/products/ps379/c1122/c dccont_0900aecd800add13.jpg you'll see the screens on the 7905/7912 (bottom left two) are substantially smaller than the 7940/7960 (top right three). So for a similar pixel size (if that is the case) the resolution is higher. By the way, the 7940/60 at least use a proportional font. It is my understanding that the 7905/12 have a smaller screen as they cannot display XML services and do not have line appearances. David Scott Laird wrote: On Mar 22, 2004, at 10:48 AM, Walker Haddock wrote: On Mon, Mar 22, 2004 at 10:16:24AM -0800, Scott Laird wrote: Can someone with a 7905 jump in? The wiki says that the display on the 7905 is higher resolution then the 7960, but doesn't give any details. How many lines of text does it show, and does it use a 1-line or 2-line format for directory entries? I'd like to order a phone today, but I'd like a bit more information first. I just went and dug around www.cisco.com. Their `data sheets` for the 7905G do not even give that information. There may be some more detailed technical documents on their site. Take a look. With Google's help, I tracked down a Powerpoint presentation that says that the 7905 is 192x64 pixels, and the 7960 is 23x9 characters. Assuming an 8x8 character matrix, the 7905 should be 24x8. I'm not sure how you can call that even higher resolution then the 7960 (http://www.voip-info.org/tiki-index.php? page=Asterisk%20phone%20cisco%2079xx). Maybe the 7905's layout is more compact? At this point, I think I've pretty much exhausted what Google and Cisco.com have to say on the topic. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VM: Multilanguage and digits
Hi! I'm trying to record som voiveprompts, and I've created a directory se in /var/lib/asterisk/sounds - in that directory I've put files like vm-intro.gsm, vm-the-person.gsm and do on. And if I use SetLanguage(se) I hear my own voice prompts! But wehre should I place the digits I've recorded? - I have tried to put them in se/digits but that don't works? Anyone that have an idea about where to put the digits for my own languages - I have read the mult-language section at voip-info.org but that don't really says where to put the sound files. Thanks for any advice! /Lars --- Lars Fredriksson Ockelbo, Sweden mailto:[EMAIL PROTECTED] http://www.fredriksson.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VM: Multilanguage and digits
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Olle E. Johansson Sent: Monday, February 23, 2004 9:41 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VM: Multilanguage and digits Lars Fredriksson wrote: Hi! I'm trying to record som voiveprompts, and I've created a directory se in /var/lib/asterisk/sounds - in that directory I've put files like vm-intro.gsm, vm-the-person.gsm and do on. And if I use SetLanguage(se) I hear my own voice prompts! But wehre should I place the digits I've recorded? - I have tried to put them in se/digits but that don't works? Anyone that have an idea about where to put the digits for my own languages - I have read the mult-language section at voip-info.org but that don't really says where to put the sound files. Hej Lars! From the Wiki: (The sound files page) Location of the sound files Asterisk normally looks for a sound file with an extension used for the codec used. If a language is set for the channel with the SetLanguage() application, Asterisk first looks for countrycode/filename where countrycode is the language code (example:. 'fr' for french). Languages and special tones for that country or region are defined in indications.conf. --- Well, this doesn't apply for digits because the source file is patched for english for some reason. The other day, this was removed for norway. Could be done for se as well, don't you agree? From say.c: /* Use english numbers if a given language is supported. */ /* As a special case, Norwegian has the same numerical grammar as English */ if (strcasecmp(language, no)) language = en; Change no to se (who cares about norwegian :-) ) and you'll be ok. And remember to report this to bugs.digium.com - tack! Hi Olle / Hej Olle! Thanks for your answer, but I don't know if I'm doing something wrong because it doesn't make any difference if I change no to se - I'm not a programmer, but I can't see how it should make any difference? Well, I solved it for the moment by replacing the digits with my swedish digits - and I will report it to bugs.digium.com - I think that might be the right way! Thanks! Best regards / Ha det gott Lars __ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Festival: read text from external fil
Hello! I wan't to use Festival for reading text from an external textfile - anyone that has a solution for doing that? I can't figure out how I should be able to do that - if it is possible? The textfile contains the temperature and will change every tenth minute - and therefore I can't use include in extensions.conf. Best regards, Lars --- Lars Fredriksson Ockelbo, Sweden mailto:[EMAIL PROTECTED] http://www.fredriksson.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming SIP-calls and Festival
Hi! I have problems with calls that are coming from a SIP-provider, and where I want to use Festival to play som text to the caller. I hear the text if I call from a SIP-extension (I've tried with g.711a/u and GSM and all three works) But if I call in to the server through my SIP-provider I wont hear any Festival-speech (no error output on the console - see in the end of the mail), if I instead use Background for example I can hear the soundfile. I think it's very strange - is there anyone that have an idea why I can't use Festival with the calls coming from my SIP-provider. This is how it looks on the console - but the caller don't hear anything; --SNIP-- -- Executing Answer(SIP/11292-594f, ) in new stack -- Executing Festival(SIP/11292-594f, 'Hello') in new stack == Parsing '/etc/asterisk/festival.conf': Found == Spawn extension (digisip, 301, 2) exited non-zero on 'SIP/11292-594f' --SNAP-- Regards, Lars --- Lars Fredriksson Ockelbo, Sweden mailto:[EMAIL PROTECTED] http://www.fredriksson.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Groups in *
Why not just use appqueue? Is that the integrated quesolution that I config in queue.conf? But as I've understood it might be a little tricky to get the users the possibility to log in/out of groups in an easy way (each extension will maybe be the member of up to four groups, and it must be possible to log in/out of each group ...) Then I need the possibility to reroute the call to another group when either all memebers of the group are busy, or when free members in the group don't answer the call for xxx seconds? Is that possible with appqueue? Okay, the solution isn't maybe the best, but it's what the customer wants ... Regards, Lars Fredriksson Hi! Thanks for the tip! Okay, looked a little around AGI and it didn't look to hard doing a script that read which phones that should answer which group from an external textfile, and such file would be quite easy to modify with a CGI-script. And I tried it with a static extensions.conf like below and it seems to work, great! Is there any other considerations or tips about using a solution like this? --SNIP from extensions.conf-- exten = s,1,Answer ; Answer exten = s,2,Dial(Sip/7101Sip/7102,20,m) ; Dial 20 seconds, if busy exten = s,103,Goto(s,3) ; go direct to next group exten = s,3,Dial(Sip/7103Sip/7104,20,m) ; Dial 20 seconds, if busy exten = s,104,Goto(s,4) ; go direct to next group exten = s,4,Dial(Sip/7105Sip/7106,20,m) ; Dial 20 seconds exten = s,5,Goto(s,2); Still no answer, goto ; first group --SNAP-- Regards, Lars - Lars: Anything you want is possible to do with Asterisk... the matter is how much time you want to spend to build that applications... I think that is posible to do that with AGI scripts... Regards, Gus - Original Message - From: Lars Fredriksson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 27, 2003 4:52 PM Subject: [Asterisk-Users] Groups in * Hi list! I have a little question about groups and Asterisk ... is there anyone out there that can say if Asterisk can do any of this; We have a customer that want call handling we cant give him with a traditional PBX, and I'm running Asterisk @home so I thought I could give it a try ... The customer wants that incoming call should go to one group with some phones in it, if the group is busy tha call should stay there for xxx seconds before it goes to another group. But if there are phones free in the group they should ring for xxx seconds before the call goes to another group. And like this it would go on with lots of groups ;-) He also wants queue messages in all groups and the possibility for the phones to log in and out of the different groups (in the morning one phone should be member of three groups, and after lunch log out of those groups and log on to another group ...) I think some kind of web-frontend would be quite kewl, so each employee could log on to a webpage and mark which groups he will answer on (I don't know how * keeps track of such things?) We have tried with PBX's like Panasonic TDA, Ericsson BusinessPhone, Avaya INDeX, Avaya IPOffice and Siemens and none of those can do this ... Thanks for any answer! Best regards Lars Fredriksson, Sweden ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __--__-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --__--__-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Groups in *
Hi! Thanks for the tip! Okay, looked a little around AGI and it didn't look to hard doing a script that read which phones that should answer which group from an external textfile, and such file would be quite easy to modify with a CGI-script. And I tried it with a static extensions.conf like below and it seems to work, great! Is there any other considerations or tips about using a solution like this? --SNIP from extensions.conf-- exten = s,1,Answer ; Answer exten = s,2,Dial(Sip/7101Sip/7102,20,m) ; Dial 20 seconds, if busy exten = s,103,Goto(s,3) ; go direct to next group exten = s,3,Dial(Sip/7103Sip/7104,20,m) ; Dial 20 seconds, if busy exten = s,104,Goto(s,4) ; go direct to next group exten = s,4,Dial(Sip/7105Sip/7106,20,m) ; Dial 20 seconds exten = s,5,Goto(s,2); Still no answer, goto ; first group --SNAP-- Regards, Lars - Lars: Anything you want is possible to do with Asterisk... the matter is how much time you want to spend to build that applications... I think that is posible to do that with AGI scripts... Regards, Gus - Original Message - From: Lars Fredriksson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 27, 2003 4:52 PM Subject: [Asterisk-Users] Groups in * Hi list! I have a little question about groups and Asterisk ... is there anyone out there that can say if Asterisk can do any of this; We have a customer that want call handling we cant give him with a traditional PBX, and I'm running Asterisk @home so I thought I could give it a try ... The customer wants that incoming call should go to one group with some phones in it, if the group is busy tha call should stay there for xxx seconds before it goes to another group. But if there are phones free in the group they should ring for xxx seconds before the call goes to another group. And like this it would go on with lots of groups ;-) He also wants queue messages in all groups and the possibility for the phones to log in and out of the different groups (in the morning one phone should be member of three groups, and after lunch log out of those groups and log on to another group ...) I think some kind of web-frontend would be quite kewl, so each employee could log on to a webpage and mark which groups he will answer on (I don't know how * keeps track of such things?) We have tried with PBX's like Panasonic TDA, Ericsson BusinessPhone, Avaya INDeX, Avaya IPOffice and Siemens and none of those can do this ... Thanks for any answer! Best regards Lars Fredriksson, Sweden ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --__--__-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channelbanks for use in europe (Sweden)
Hi! Is there anyone that are using a E1-channelbank and have any tips about some type? Im looking at the TE410P and use one port for a PRI (Euro-ISDN, I think we're using some slightly modified version here in Sweden, but I'll check that tomorrow) and connect one port to a channelbank for 30 analogue telephones. It would also be great to get callerid on the analogue phones, so it would be intresting to know how the channelbank sends the callerid (DTMF or FSK) if it sends any callerid? Best regards, Lars Fredriksson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Groups in *
Hi list! I have a little question about groups and Asterisk ... is there anyone out there that can say if Asterisk can do any of this; We have a customer that want call handling we cant give him with a traditional PBX, and I'm running Asterisk @home so I thought I could give it a try ... The customer wants that incoming call should go to one group with some phones in it, if the group is busy tha call should stay there for xxx seconds before it goes to another group. But if there are phones free in the group they should ring for xxx seconds before the call goes to another group. And like this it would go on with lots of groups ;-) He also wants queue messages in all groups and the possibility for the phones to log in and out of the different groups (in the morning one phone should be member of three groups, and after lunch log out of those groups and log on to another group ...) I think some kind of web-frontend would be quite kewl, so each employee could log on to a webpage and mark which groups he will answer on (I don't know how * keeps track of such things?) We have tried with PBX's like Panasonic TDA, Ericsson BusinessPhone, Avaya INDeX, Avaya IPOffice and Siemens and none of those can do this ... Thanks for any answer! Best regards Lars Fredriksson, Sweden ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP, X-Lite
Hi folks! I bought a X100P a while ago and know I've tried to get it working here at home again ... but I can't manage to get my X-Lite client working with Asterisk (CVS from a day ago) ... I've downloaded the latest version of X-Lite and I believe that I've set it up correctly ;-) But I cant get it to register with my Asterisk - I only get Login timed out, contact your network admin But, I can call voicemail and other SIP clients anyway - I can call voicemail and other SIP clients even if I enter a username that is not existing in my sip.conf??? The only error message I get in my Asterisk console is; NOTICE[81926]: File chan_sip.c, Line 5119 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.5.10' 192.168.5.1 is the Asterisk server and 192.168.5.10 is my client. Below is my sip.conf - is there anyone that can ponit out what I've done wrong I would be very, very, very happy ;-) Maybe an short description in what I would enter where in the X-Lite configuration wouldn't hurt ... Thanks for any help! Best regards Lars Fredriksson, Sweden [general] port = 5060 ; Port to bind to bindaddr = 192.168.5.1 ; Address to bind to context = default ; Default for incoming calls ;srvlookup = yes; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registrati ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video ;disallow=all ; Disallow all codecs ;allow=ulaw ; Allow codecs in order of preference ;allow=ilbc ; ;register = [EMAIL PROTECTED] ; Register with a SIP provider ;register = [EMAIL PROTECTED]/1234 ; Register 2345 at sip provider as 123 ; [sip7101] context=sip type=friend secret=blah auth=md5 ; defaultip=192.168.5.10 host=dynamic dtmfmode=inband mailbox=7101 -- Using M2, Opera's revolutionary e-mail client: http://www.opera.com/m2/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P in Sweden / Europe
Hello folks! I've been playing with Asterisk home for a while now (just internal with IAX/SIP-clients). But I think it would be nice to try interface it with the real world. And I'm wondering if someone tried the X100P in Sweden or Europe at least, or if someone knows if it should work here ;-) Thanks! Best regards, Lars Fredriksson - Ockelbo, Sweden ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
