similar to: How To Separate incoming channels from the others on a PRI

Okay, here is the background. I have a PRI with 15 active channels on it. I originally setup all of them in group=1 and all outgoing and incoming calls used this group. The phone number that I have associated with these channels ends with 750 and that is how I direct the calls. i.e. In my extensions.conf I have: exten => 750,1,Dial(SIP/120,20) All this works fine. Now I have the need

Okay, here is the background. I have a PRI with 15 active channels on it. I originally setup all of them in group=1 and all outgoing and incoming calls used this group. The phone number that I have associated with these channels ends with 750 and that is how I direct the calls. i.e. In my extensions.conf I have: exten => 750,1,Dial(SIP/120,20) All this works fine. Now I have the need

I have an extension setup in my extensions.conf for hold music. ext. 600. If I pick up a phone (polycom soundpoint 300 sip) and dial extension 600 I hear the hold music playing. If I call another extension and pick it up and put the call on hold with the hold button on the phone I hear nothing at all. Does anyone have any experience with these phones and getting the hold button to work?

I've read as much as I can on this error and still can't seem to figure out what's causing this error: Jun 23 20:54:58 NOTICE[7483]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 23 20:55:03 NOTICE[7483]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 23 20:55:03 NOTICE[7483]:

Hello all! We have an E400P on a dual Xeon box with four EuroISDN PRI to the PSTN and Cisco ATA-186 as SIP UA. We experience very large amount of echo (on -some- calls) when we're doing PRI->SIP calls but not when doing SIP->PRI calls. We don't think the problem is IP latency related (When calls are made PRI->*->PRI we still experience the echo). What do you think? Regards

Hello all! We have an E400P on a dual Xeon box with four EuroISDN PRI to the PSTN and Cisco ATA-186 as SIP UA. We experience very large amount of echo (on -some- calls) when we're doing PRI->SIP calls but not when doing SIP->PRI calls. We don't think the problem is IP latency related (When calls are made PRI->*->PRI we still experience the echo). What do you think?

Hi everyone, We have a Asterisk-based call center deployment with around 40 SIP users, attending incoming calls from two PRI lines (2xE1) using agents and queues. The problem is that Asterisk stops accepting new incoming calls to the PRI lines without reason, although there should be free channels to make room for new incoming calls, but Asterisk thinks these channels are being used. SIP calls

I have a single PRI to our local telco that is shared by multiple companies. I would like to find a way to get our long distance carrier to provide separate billing for these companies. I am currently calling Authenticate() to collect an account code which is written to the call detail record, then I estimate the charges for each company by parsing the call detail records. This doesn't work

Good morning, I have a Bell Canada PRI here (switchtype=national) and I am trying to perform a call-forward-unconditional on one of the DIDs. The idea is that when DID 5551234 receives a call, Asterisk redirects it back out the same PRI to some external number. This is simple enough to do with something along these lines: [PRI] exten => 5551234,1,Set(CALLERID(RDNIS)=${EXTEN}) exten =>

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Adams, Gavin Is there any additional information I could provide to start tracking this down? I was thinking about looking into the various applications source to see how they access the data elements for callerid. I know where the values are pulled

Hi, I am using Asterisk with various brands and models of SIP phones. Especially the Welltech phones LP201 are particularly nasty with volume and echo. Even with the input gain (microphone) of the Welltech set to the max, the PSTN end can hardly hear the SIP user on incoming calls. Ztmonitor also only gives a level of around 3 === from the SIP phone. I have to increase the rxgain and txgain

HI, I'd like to redirect an incoming E1 call to a local landline, at the moment I just do Exten => thenumber,1,Dial(Zap/g1/localnumber) However this seems to cause all sorts of problems with the fax machine on the end of that landline. Is there a better way to redirect a call? Cheers, Ben -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, whatever dialplan I'm using for outgoing calls via PRI (Digium card, chan_zap), the callerid when receiving calls has no leading zeros, which are normally used to distinguish local, national and international calls in Europe. The number has always the area code in front, but the country code only for foreign calls. Now I'm looking for any mean to decide, whether the received

Hi, i managed to connect to Ericsson MD110 with PRI at last. And made a successful call thru asterisk to ericsson. But when i try to call from ericsson to asterisk i got an error on asterisk side. And i couldnt figure out why. Here's my extensions.conf about incoming calls. [DID_span_1] include = DID_span_1_timeinterval_all,${timeinterval_all} DID_span_1_timeinterval_all] exten =

Relax your PRI is fine. What Xo is sending you is 4 digits os DID. If for example you have 1130-1153 as the last 4 digits of your Number you can use this to rout your calls. exten => 1130,1,Goto(ivr,s,1) Exten => 1140,1,Goto(extensionss,100,1) Exten => _X.,1,Zapateller Is the above config. XXX-XXX-1130 with go to the s extensions' first priority in the ivr context. XXX-XXX-1140

I just got a TE110P up on an XO PRI - everything looks good so far. We've been given a block of 23 numbers for the PRI. If I explictly set the incoming extension in extensions.conf like: exten => 1153,1,Answer or: exten => _XXXX,1,Answer I can get the incoming call. If I try and do: exten => s,1,Answer I'll see something like this: -- Extension '1153' in context

Thanks James, What i need is to make the fax machines connected to the audiocodes mediant 1000 be able to send and receive fax throught Asterisk (connected to a pri) I know it's not reliable, but it should work at leaste, what should i do on Asterisk and Mediant to make this work? Im quite confuse with all these fax issues :S Thanks in advance > > Message: 11 > Date: Fri, 9 Apr

Hey all, We've upgraded our PRI trunk to support what BellSouth calls "enhanced caller id name delivery". The weird part is, I'm only capable of seeing the name portion on incoming calls within voicemail2's email delivery. For example, on an incoming call, asterisk is reporting this: Context from extensions.conf (BS delivers 7-digit DIDs): exten => 9133727,1,Answer

Was'nt sure if this mail got through earlier: I have been having a weird issue with my telco's ISDN PRI occasionally resetting on a incoming call, i suspect it to possibly be a timing issue since some of the incoming call work. This problem happens very frequently. I am using asterisk-1.6.0.1 with libpri-1.4.9, the asterisk server is connected viw TDMoE to a Redfone Fonebridge into

Hi all, I have a E100P digium card installed in my server, and now I am working with the incoming call to this PRI interface. my extensions.conf file for incoming call is: [default] exten => , 1,Answer exten => s,1,Dial(SIP/2001,20) exten => s,2,Hangup And when the incoming call comes, it appears: -- Executing Dial("Zap/1-1", "SIP/2001|20") in new stack