similar to: Powertouch 350 CallID display continued

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

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

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

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

Hi All... Well, I'm still experiencing LOTS of dropped calls since installing the new (non pri) T1 here... I keep noticing a few things in the logs when this happens, namely the "Wink/Flash" statements and the "Didn't get a frame" messages... Anyone got any ideas on if this is a telco issue, a wiring issue, or an asterisk issue? Been trying to track this down via all 3

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

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, To combine two sources of CDR's I want Asterisk to save the SIP callid for all calls. I know there's a variable that contains the SIP CallID value, but is this the callid value of the incoming INVITE message or the outgoing message? Are they the same? (I've not yet checked a trace, I'm sorry for that). I've tried to read chan_sip, but couldn't find something in the

Anyone gotten two of the 2572 echo canceller cards to work in a 253c mounting assembly? I can get one to work, but when I install two, one always fails. I've tried all my cards solo in the enclosure, on each side, and they all work properly when only 1 is installed, however, when I install two, one of them will come up, but the other always fails. Anyone know what might be causing this?

Hello everyone Is there a possibility to catch an outbound callleg ID for the follovong scenario: some carrier -----> ------(asterisk1) --->-----asterisk2 ? I can get inbound callid for asterisk1 with a ${SIPCALLID} in extensions.conf or to look it up in cdrs field (are the same). But how about outbound? I have all calls just forwarded through asterisk1, not answered and for every call I

Hello all, I tried the dev list, but got no answer at all. I'm facing some problems with call-id generation in a heavily loaded Asterisk Server. Asterisk is generating same call-id and from tag for different calls (and this is not desirable). Looking at the source code I noticed that rand() is used four times to get a callid. Is that safe enough? Maybe my system lacks of a good random

30 says it's view only in the docs & I can't seem to change it, any other options? > Option 30 allows to set Module Shelf Address/ID.

Following up with some more tests... I just left a message in a voicemail box, and got the same behaviour (i.e. it says its starting to play ('First Message') then it goes to the VM ending menu (i.e. press 7 to delete) without playing the message. On this particular test.. I went ahead & left a 2nd VM in the inbox, then when I went to check them, they would both play properly... The

Hi, I am making calls from the dial plan using the dial application. Due to technical requirements I need to find out the sip call-id used in the dialog initiated by the dial application. I dont see any straight forward way of doing this so I am looking for answers. There is a sip callid session variable but the problems is that dial is a blocking call and the dialog ends when dial returns. I

I haven't intentionally done anything to the CID, the odd thing is that another aastra phone (480e) connected to the same port DOES display CID, it's only these PT350s that can't seem to read it.. Been talking to Aastra support & they've suggested some avenues to explore, but none of them have worked... The PT350s definitely have a problem w/*'s cid signals.. Just not sure

I am not getting any caller Id with my standard T-1. Is a standard "T" capable of sending callerid? I don't want to spend time troubleshooting my PBX if Asterisk can't send it down that type of trunk. Jordan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Calling linphone from asterisk 13.9.1.: Dial(SIP/<user>@sip.linphone.org) And it works. But on the linphone side the caller is: <extno>@ipaddress or 2502 at 45.123.987.4 Is there any way to make it more descriptive, at least for the sip user name ? I tried setting SIPCALLID, which had no effect. Set(SIPCALLID=Office) Thanks, sean

Hey all, been noticing some oddness on a new AAH install... occasionally an incoming zap line with automatically connect with an outgoing extension, even though the incoming line hasn't specified what extension it's aiming for (i.e. haven't tapped in the ext # yet)... so someone's trying to call out from inside the office & are automatically connected with an incoming line.

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 ;

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