similar to: RE: Asterisk-Users Digest, Vol 22, Issue 132

Hi All, posted last week but didn't get any responses. I'm trying to figure out why some of our analog phones aren't showing CID when hooked up to asterisk. To recap, I have an Aastra Powertouch 350, which shows caller ID fine when connected to the PSTN, but when hooked up to asterisk, CID does not show. I've hooked up another phone to the same * port that the Aastra phone is on,

Hi all, is anyone using the Aastra Powertouch 350 analog (adsi) phone with asterisk? I cannot get the phone to display incoming caller id... I can see the CID if I hook up a cheap caller id enabled phone... Only the PT350 is having problems. Looking all over for docs on this phone, but what I've found doesn't have any mention of a setting to display CID... Thx as always Dan

Hello, I'm about to install asterisk as the PBX at a location that my company has just moved into and I would like to get some comments and advice on the installation. I am new to * and don't want to make any big mistakes so I would love to hear whatever anyone has to say. Here is what I have so far Server: * 2.8Ghz P4 - 1G ram * T400P Tormenta II (is this as good as the

Program your locked ADSI phone not by unlocking it but by using the feature slot FDN download descriptors of ADSI SEC security code telco provider lock number key from database list. ch33s3 First reset phone. Set clock to Jan 1 12:00AM (options-2?). exit to main screen. press options, (mute/flash/speaker/?) until you see firmware version. press # to reset. google more info, bam ch33s3d ;

I just need to buy 5-6 ADSI Phones. I wondering which of these models to choose - aastra 390 - I don't know is this an ADSI phone at all. Is there versions with and without ADSI - aastra 350 - I'm sure that it have ADSI. If there is some other good working model, it will be great, if someone points me. Thanks

Hi All, as a followup to my previous posting (re: not getting caller id to display on a powertouch 350), I've found the following... If I hook the PT350 up to the PSTN line, caller id is diplayed properly... If I connect it to our CAS Acess Bank II, the 350 will not display incoming CID... If I hook up a cheap callerid enabled phone to the same port that the PT350 was on (on the AB II), the

hello i hope i am posting to the right list, i am a totally blind user, and i want to reprogram my aastra pt 480e phone, my friend used the web configurator, but i think he programmed thw wrong codes, a few questions, is it possible to damage the phone by programming it wrong? also, how does one reset it literally to factory defaults? as unplugging it and replugging it back into the wall

I am having a problem trying to do inband DTMF passthru via asterisk. My setup: PSTN gateway MAXTNT v11.0 SIP (T1 PRI/NT2) Asterisk HEAD or v1.0 makes no difference (I am using HEAD mostly) 12/10/04 and 01/17/05 (no difference) CAC ABII-T100P to/from analog lines to/from asterisk BTW, I have used a ABI and it works just like the ABII with asterisk. What I am seeing is: I make a call from a

Hello list, I have reason to believe that ADSI can be spoken to phones even when they're onhook. Is this true? Does anyone know? Right now, I'm having trouble figuring out how to do anything to an onhook channel other than ring it. Does this require a magic application and some serious voodoo? Any pointers? I got onto this idea because I noticed that whenever I got voice mail on a

There isn't much documentation on adsi, but I called NETXUSA (the vendor of my 480e) and they helped me along. My experience: 1. I really had no experience with ADSI so I had (probably still have) some misconceptions on how the configuration is loaded onto the phone. 2. I set the following in my /etc/asterisk/asterisk.adsi (most of this is the stock asterisk.adsi script): ;

Hi, Would anyone know of a way to set the time automatically on an ADSI capable phone from *? The phone in question is a Aastra 480e. While I am at it, does anyone have any helpful docs on the ADSI script programming? I have managed to do basic functions by modifying the asterisk.adsi file using stuff gleaned from the app_adsiprog.c file, but docs would be really helpful at this point. Tia,

I was wondering if anyone has had any experience with PowerTouch 350 & ADSI. I am able to configure the phone with ADSI.. Which is pretty cool in it self. But I have a question regarding the Service feature. I have to choose the service button and choose the Asterisk PBX ADSI. But let's say I make a call it does not default back to this. I have to go back into Services and choose it

Hi all, I'm getting some noise gate like effects on our sip lines & I think I need to disable silence supression, I'm searching docs & not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd

I have a question hoping someone can help me with. I just purchased a 350 from http://lktelecom.zoovy.com/product/HPT350. I have been reading some posts and heard that these phones won't work with Asterisk.. Is this true? I had a previous Nortel Phone working in this enviroment. Well anyway here's my problem. The phone keeps ringing.. It won't stop.. Which makes the phone

Recently, I've been having issues with the URIs returned from e.164.org and toll free calls. It seems that the URIs that are returned from ENUMQUERY and ENUMRESULT are no longer the proper numbering schemes that the poviders use. I've been using the following [enum] template in my outbound route for quite some time with great success until recently. [enum](!) exten =>

Hi again, I'm trying to setup our non pri T1 (they call it a Long Distance T1), our current pbx has the signaling set to E&M, I can set em in zapata.conf, but I'm trying to track down the proper entries for the zaptel.conf file. The digium docs only show a PRI example. Our current system has these settings: Signalling: E&M Framing mode: ESF Line Coding: B8SZ here's my

thank you for your response below the asterisk -vvvr Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0176XXXXXX at from-internal:1] Macro("SIP/300-00000192", "user-callerid,LIMIT,EXTERNAL,") in new stack -- Executing [s at macro-user-callerid:1] Set("SIP/300-00000192", "TOUCH_MONITOR=1427481319.470") in new stack --

Hi, I can't find a way to tailor DHCP/TFTP/HTTP environment so that brand new Aastra SIP phones can be auto-provisioned when config files are stored in a specific TFTP subdirectory instead of TFTP root directory. For instance, TFTP root directory is /srv/tftp. When config files are stored in /srv/tftp, a new Aastra can find its config files. When config files are stored in /srv/tftp/aastra,

Hi, What are the main differences between Aastra SIP phones 6730i and 6750i series ? Aastra corporate web site mentions : "The Aastra 6730i Series offers exceptional features and flexibility in an open-standard enterprise grade IP telephone" for one "The Aastra 6750i Series offers features and flexibility in an open-standards based, carrier grade IP telephone." for the

If you own Aastra phones, here's a group dedicated to your specific needs. BTW - The Asterisk-users mailing list is great but it has way too much volume to be useful for specific problems. It needs to be broken up into smaller more manageable lists. Homepage: http://groups.google.com/group/aastra-asterisk-users Group email: aastra-asterisk-users@googlegroups.com