[asterisk-users] Continued NTL ISDN issues with Vox D210E card
Hi Im getting no further will getting a Openvox D210E card connected to an NTL/Virgin Media UK PRI connection. Wondering if any of the following information might help? If I try and dial I will get [May 22 19:58:13] WARNING[4087]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'ZAP' (cause 0 - Unknown) I get various errors in the asterisk cli: == Primary D-Channel on span 1 down [May 22 19:51:32] WARNING[3749]: chan_zap.c:2402 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! == Primary D-Channel on span 1 up Some reboots will give [May 22 20:29:49] ERROR[3884]: chan_zap.c:8248 zt_pri_error: !! Got a UA, but i'm in state 7 Pri Debugs: q921.c:777 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:728 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED Sending Set Asynchronous Balanced Mode Extended repeated constantly -- Got SABME from network peer. Sending Unnumbered Acknowledgement I can 'sometimes' get the system to work on a reboot, maybe once in 15-20 times, then it fails again. If I play around with the zaptel.conf and get it wrong so the zaptel service doesn't start, re-edit, then start zaptel and then load asterisk it works. Currently running Asterisk 1.4.20 and Zap 1.4.10.1 on Suse 10.2 Ive had to used the /etc/init/zaptel script fix to allow it to run on Suse (http://users.otenet.gr/~becos/zaptel.init.txt) Only thing I havent tried so far is dropping back to Asterisk 1.2 or an earlier 1.4 Zaptel: # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 span=1,1,1,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 span=2,2,1,ccs,hdb3 # termtype: te bchan=32-46,48-62 dchan=47 # Global data loadzone = uk defaultzone = uk Zapata [trunkgroups] [channels] context=ntlisdn30in switchtype=euroisdn pridialplan=local prilocaldialplan=local signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaiting callerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no callprogress=no callerid=asreceived group=1 signalling=pri_cpe channel = 1-15,17-31 group=2 signalling=pri_cpe channel = 32-46,48-62 Tim This message is sent in confidence for the addressee only. Unless specifically stated, the contents are not to be disclosed to anyone other than the addressee. Unauthorised recipients must preserve this confidentiality and should please advise the sender immediately of any error in transmission. The views an opinions expressed in this e-mail message are the sender's own and do not necessarily represent the views and opinions of NS Optimum Ltd. Although this e-mail and attachments are believed to be free of any virus or other defects which may affect any computer or IT systems into which they are received, no responsibility is accepted by NS Optimum Ltd for any loss or damage arising in any way from the receipt or use thereof. Place of registration: England, Registered Office: Jenton Road, Sydenham Ind Est, Leamington Spa, Warwickshire, CV31 1XS, Registered No 3018839 ___ -- 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] Out-Going Calleriid
Thanks for the heads up again guys. Still no go. It's a ISDN30 PRI on NTL(Virgin) in the UK I currently have a Mitel 3300 connected happily sending CallerID's so I know it the teleco supports it. The Mitel is set to send 01926xx so that's what I'm trying to get Asterisk to send. Running an Openvox D210E that runs with wct4xxp drivers. It definitely work before so it must be something I've done, OR, a certain driver / zaptel version Caller id in-coming is fine, just won't send out. Huff Tim p.s Sorry for the disclaimer. Should be gone on this one. This message is sent in confidence for the addressee only. Unless specifically stated, the contents are not to be disclosed to anyone other than the addressee. Unauthorised recipients must preserve this confidentiality and should please advise the sender immediately of any error in transmission. The views an opinions expressed in this e-mail message are the sender's own and do not necessarily represent the views and opinions of NS Optimum Ltd. Although this e-mail and attachments are believed to be free of any virus or other defects which may affect any computer or IT systems into which they are received, no responsibility is accepted by NS Optimum Ltd for any loss or damage arising in any way from the receipt or use thereof. Place of registration: England, Registered Office: Jenton Road, Sydenham Ind Est, Leamington Spa, Warwickshire, CV31 1XS, Registered No 3018839 ___ -- 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] Out-Going Callerid
Well.. Now I'm confused. Recap.. Couldn't appear to get out going callerid to work on a UK NTL PRI connection. Id been testing it with my Orange Mobile phone.. Dial the 07973xx and it displays private. Called my girlfriend tonight on our land line (all be it NTL again but this time analogue), got her to do 1471 and feck me, it read back the callerid Id been putting through. Only been able to try it on Orange and NTL residential at the moment. Ill try it to a BT line tomorrow morning. I'm really stumped now.. Why does it work on one and not on the other? Tim This message is sent in confidence for the addressee only. Unless specifically stated, the contents are not to be disclosed to anyone other than the addressee. Unauthorised recipients must preserve this confidentiality and should please advise the sender immediately of any error in transmission. The views an opinions expressed in this e-mail message are the sender's own and do not necessarily represent the views and opinions of NS Optimum Ltd. Although this e-mail and attachments are believed to be free of any virus or other defects which may affect any computer or IT systems into which they are received, no responsibility is accepted by NS Optimum Ltd for any loss or damage arising in any way from the receipt or use thereof. Place of registration: England, Registered Office: Jenton Road, Sydenham Ind Est, Leamington Spa, Warwickshire, CV31 1XS, Registered No 3018839 ___ -- 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] Out-Going Callerid
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Guy Sent: 08 May 2008 22:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Out-Going Callerid Well.. Now I'm confused. Hmm.. Just phoned the sprogs mobile of O2 and its still withheld. This message is sent in confidence for the addressee only. Unless specifically stated, the contents are not to be disclosed to anyone other than the addressee. Unauthorised recipients must preserve this confidentiality and should please advise the sender immediately of any error in transmission. The views an opinions expressed in this e-mail message are the sender's own and do not necessarily represent the views and opinions of NS Optimum Ltd. Although this e-mail and attachments are believed to be free of any virus or other defects which may affect any computer or IT systems into which they are received, no responsibility is accepted by NS Optimum Ltd for any loss or damage arising in any way from the receipt or use thereof. Place of registration: England, Registered Office: Jenton Road, Sydenham Ind Est, Leamington Spa, Warwickshire, CV31 1XS, Registered No 3018839 ___ -- 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] Out-Going Calleriid
Installing a new box onto UK NTL (Virgin Media)
During testing phase the callerid worked, now it doesn't.
Can someone confirm that my syntax is right before I start ripping the
configs to bits
exten = _9.,1,Set(CALLERID(number)=01926xx)
exten = _9.,2,Dial(ZAP/1/${EXTEN:1})
Ive tried all permutations of the CALLERID (ie CALLERID(NAME) and
CALLERID(NUMBER) but it just wont work anymore.
Zapata has the following relevant settings
usecallerid=yes
hidecallerid=no
callwaiting=yes
Im Stumped
Tim
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Re: [asterisk-users] Out-Going Calleriid
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: 07 May 2008 20:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Out-Going Calleriid The leading 0 is not part of Caller*ID. Remove it. Thanks for your mail Eric. Its not that Im afraid. Dialing my mobile I'm still getting 'Private Caller' This message is sent in confidence for the addressee only. Unless specifically stated, the contents are not to be disclosed to anyone other than the addressee. Unauthorised recipients must preserve this confidentiality and should please advise the sender immediately of any error in transmission. The views an opinions expressed in this e-mail message are the sender's own and do not necessarily represent the views and opinions of NS Optimum Ltd. Although this e-mail and attachments are believed to be free of any virus or other defects which may affect any computer or IT systems into which they are received, no responsibility is accepted by NS Optimum Ltd for any loss or damage arising in any way from the receipt or use thereof. Place of registration: England, Registered Office: Jenton Road, Sydenham Ind Est, Leamington Spa, Warwickshire, CV31 1XS, Registered No 3018839 ___ -- 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] Was; Multiple Hunt Group to same extension advice Now: CallerID insert
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Guy
Sent: 30 April 2008 18:22
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Multiple Hunt Group to same extension advice
Swapping out our old Mitel 3300 for Asterisk I need to come up with a
solution to advertise to the extension what hunt group the call was
for,
plus distinguishing those calls from calls that are sent straight to
the
extension.
OK I answered my own question. Use CALLERID NAME in the exten = except
that the Grandstream 1200 GXP I have on test will only disable the
CALLERID NAME not the NUMBER. Well it certainly does from extension to
extension, I currently can not hook it up to the PSTN
The Zoiper software phone does its job and displays both.
So I used exten = xxx,n,Set(CALLERID(name)=Technical:${CALLERID(name)})
Cracking
Cheers
Tim
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[asterisk-users] Queues v Multi Device dialing
Creating some 'hunt groups' Im not sure if to go for queues or just dial multi devices (one group is around 15 handsets) I don't need the queue features, although I am backgrounding thank you for holding every 30 seconds. Does one solution create more processor / networking overheads than another, or the fact I'm ringing the same amount of phones in both solutions mean they are equal? Cheers Tim This message is sent in confidence for the addressee only. Unless specifically stated, the contents are not to be disclosed to anyone other than the addressee. Unauthorised recipients must preserve this confidentiality and should please advise the sender immediately of any error in transmission. The views an opinions expressed in this e-mail message are the sender's own and do not necessarily represent the views and opinions of NS Optimum Ltd. Although this e-mail and attachments are believed to be free of any virus or other defects which may affect any computer or IT systems into which they are received, no responsibility is accepted by NS Optimum Ltd for any loss or damage arising in any way from the receipt or use thereof. Place of registration: England, Registered Office: Jenton Road, Sydenham Ind Est, Leamington Spa, Warwickshire, CV31 1XS, Registered No 3018839 ___ -- 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] Multiple Hunt Group to same extension advice
Swapping out our old Mitel 3300 for Asterisk I need to come up with a solution to advertise to the extension what hunt group the call was for, plus distinguishing those calls from calls that are sent straight to the extension. On the Mitel I had the handset extension programmed on the primary line key, and then, as in my example, Sales calls to another key, Technical to another, and accounts to another. If the call was sales, technical, or accounts, I could ignore it unless no one picked up, and if it was direct to me I knew I had to get it. But with Asterisk, I can get the external caller to go through the auto-attendant and then get the call to direct to multiple sip phones but I don't know then if the call is really for me, or if it was just for a department. I'm still using my Mitel handsets, so maybe its just a limitation of those and I need a different type of handset? Can I put something into the caller ID that says this call is from xxx-xxx-xxx and is for technical? Confused Tim This message is sent in confidence for the addressee only. Unless specifically stated, the contents are not to be disclosed to anyone other than the addressee. Unauthorised recipients must preserve this confidentiality and should please advise the sender immediately of any error in transmission. The views an opinions expressed in this e-mail message are the sender's own and do not necessarily represent the views and opinions of NS Optimum Ltd. Although this e-mail and attachments are believed to be free of any virus or other defects which may affect any computer or IT systems into which they are received, no responsibility is accepted by NS Optimum Ltd for any loss or damage arising in any way from the receipt or use thereof. Place of registration: England, Registered Office: Jenton Road, Sydenham Ind Est, Leamington Spa, Warwickshire, CV31 1XS, Registered No 3018839 ___ -- 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] Next Move - Hosting
I posted this to asterisk biz but didn't get a reply.. I didn't want to offend anyone being that this is kind of branching into hosting, and maybe outside of the remit of this list. Hi Been lurking on the user list for a while but I have some what of an immediate requirement and I'm wondering if you can suggest the best solution (if mines a rubbish idea) I have been testing Asterisk as a bolt on to our Mitel 3300.. its been doing some softphones for users abroad, etc and I'm happy with the fact I want to progress to a full system. However during this testing phase 2 customers of mine (I'm a IT Service Provider) have ask for some managed, collocated small business servers, which include the requirement for me to host their phones. No Problem I thought, I'm well on the way to this anyhow. So I'm thinking (although not tried it) that if I got my Asterisk box running for my company (E1 card for outside link) I would AIX the hosted PBX for the customers to my PBX to allow them to make outgoing calls. I would get my teleco to provide phone numbers for them and also get my PBX to redirect that number to the hosted PBX. Is this correct so far? Or should I keep their system separate on another E1? Or should I forget my PBX and push their incoming / outing calls out to a SIP / AIX provider on the net and wash my hands of it? Also I know you can run multi context on one host BUT can they also run the same extension numbers? Or would I have to let one company have 401-410 and the next company have 411 to 420, etc, etc (I'm guessing that's the case) And lastly.. Call accounting.. Certainly found a lot of good info about certain call accounting applications but as anyone got any good feedback about one they personally use.. Id like to keep it GNU / Open Source / Free while I build myself up.. Although I don't want to compete with the big boys, Id like to think I could get 10-20 or so customers co-located. Cheers Tim This message is sent in confidence for the addressee only. Unless specifically stated, the contents are not to be disclosed to anyone other than the addressee. Unauthorised recipients must preserve this confidentiality and should please advise the sender immediately of any error in transmission. The views an opinions expressed in this e-mail message are the sender's own and do not necessarily represent the views and opinions of NS Optimum Ltd. Although this e-mail and attachments are believed to be free of any virus or other defects which may affect any computer or IT systems into which they are received, no responsibility is accepted by NS Optimum Ltd for any loss or damage arising in any way from the receipt or use thereof. Place of registration: England, Registered Office: Jenton Road, Sydenham Ind Est, Leamington Spa, Warwickshire, CV31 1XS, Registered No 3018839 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz This message is sent in confidence for the addressee only. Unless specifically stated, the contents are not to be disclosed to anyone other than the addressee. Unauthorised recipients must preserve this confidentiality and should please advise the sender immediately of any error in transmission. The views an opinions expressed in this e-mail message are the sender's own and do not necessarily represent the views and opinions of NS Optimum Ltd. Although this e-mail and attachments are believed to be free of any virus or other defects which may affect any computer or IT systems into which they are received, no responsibility is accepted by NS Optimum Ltd for any loss or damage arising in any way from the receipt or use thereof. Place of registration: England, Registered Office: Jenton Road, Sydenham Ind Est, Leamington Spa, Warwickshire, CV31 1XS, Registered No 3018839 ___ -- 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] RIM Blackberry WLAN SIP phone
Now if only Nokia or a third party could come up with a SIP client for the new Nokia 7500 Communicator. I would be a happy chappy!! Tim -Original Message- From: John Breeden [mailto:[EMAIL PROTECTED] Sent: 05 November 2004 04:59 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RIM Blackberry WLAN SIP phone Blackberry announced a sip based voip phone available early '05 http://www.blackberry.com/news/press/2004/pr-18_10_2004-02.shtml ___ 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] P32mxi
Can anyone confirm if they have this channel bank running on Asterisk? There is a post in the archive about having a few niggley problems but no follow up. I tried to email the guy but the mail address is bouncing. I'm not having much luck finding any of the archive recommended channel banks cheap in the UK. I really want to try and get someone that someone has already had success with (being a numpty). I am finding no reference to the 650 /750's that everyone is suggesting. Ill keep looking, cheers Tim ___ 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] UK install
Well I'm slowly learning my way around asterisk although as yet I haven't had the chance to actually hook the system up to an ISDN line. I am going to migrate from an Argent Office setup. My only problem is keeping costs down on the phones. The Argent system is running about 30 POTS phones. Can someone suggest the cheapest option? Should I get some kind of large scale FXS box or would the cost of doing that on a large scale work out the same as getting cheap SIP phones? I have a large number of POTS phones with headsets so I would have to take that into account if I replaced the phones with SIP's In an ideal world Id like to convert a number of POTS to soft phones but as always its persuading the users that they can operate in the same way. Our Telco is NTL offering us an ISDN 30 style package. I assume this is a E100P card requirement? Any suggestions for good UK reseller or shall I get it direct from Digium? Anyhow, as I say I'm getting more functionality out of Asterisk than I ever did with (personally thinking) a very confusing Argent setup. I just hope that I can make it financially viable to do the install Cheers Tim ___ 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] Background Playback fails
Hi Guys. Ive had a lay off from Asterisk for 12 months but I am starting to look into it again. I am not very Linux savvy and found it hard going the last time. Ive started playing with it in the last 3 weeks and I have to admit to making more head way this time. The first problem I'm stuck on and I cant find a solution to is that sound files that I have recorded (be it by creating a temp file from an extension number, or by recording in .wav) will not play when I try to create a menu in the extension.conf The relevant lines are: exten = s,1,Wait,2 exten = s,2,Answer exten = s,3,Background(mainmenu) The mainmenu.gsm is sitting in the sounds folder. I get an error on the console saying: -- Executing Wait(SIP/timg-a4e6, 2) in new stack -- Executing Answer(SIP/timg-a4e6, ) in new stack -- Executing BackGround(SIP/timg-a4e6, mainmenu) in new stack Jun 11 12:33:00 WARNING[1209214400]: file.c:464 ast_openstream: File mainmenu d oes not exist in any format Jun 11 12:33:00 WARNING[1209214400]: file.c:752 ast_streamfile: Unable to open mainmenu (format ULAW): No such file or directory == Spawn extension (default, s, 3) exited non-zero on 'SIP/timg-a4e6' What am I doing wrong guys??? Cheers Tim
RE: [Asterisk-Users] Background Playback fails
Not totally. I did read an archive that said DONT put .gsm or .wav on the end (I have to admit to trying) so I assumed paths was a no-no as well. Shall I try it?? Tim -Original Message- From: usedcanon [mailto:[EMAIL PROTECTED] Sent: 11 June 2004 12:40 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Background Playback fails have tried specifying the full path ? Umar
RE: [Asterisk-Users] Background Playback fails
It worked! Cool I assumed that as the demo sounds didnt have paths, mine wouldnt need them either. Thanks umar -Original Message- From: usedcanon [mailto:[EMAIL PROTECTED] Sent: 11 June 2004 12:40 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Background Playback fails have tried specifying the full path ? Umar
