Re: [asterisk-users] FWD and IPCall
I am using Asterisk and IPKall.http://lists.digium.com/pipermail/asterisk-users/2008-January/203607.html I tried exactly below mentioned configuration .http://lists.digium.com/pipermail/asterisk-users/2008-January/203607.html http://lists.digium.com/pipermail/asterisk-users/2008-January/203607.html But I get an Error as http://lists.digium.com/pipermail/asterisk-users/2008-January/203607.html == Parsing '/etc/asterisk/sip_notify.conf': Found -- Executing NoOp(SIP/66.54.140.46-b781b470, from-ipkall) in new stack -- Executing NoOp(SIP/66.54.140.46-b781b470, INTL. NUMBER/INTL. NUMBER 2067770020) in new stack -- Executing Dial(SIP/66.54.140.46-b781b470, Local/200 at internal) in new stack -- Called 200 at internal -- Executing AGI(Local/200 at inter...@default-f232,2, agi:// 127.0.0.1:4577/call_log) in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial(Local/200 at inter...@default-f232,2, SIP/200 at inter...@sip64||tTor) in new stack -- Called 200 at inter...@sip64 -- Local/200 at inter...@default-f232,1 is ringing Jan 15 17:18:43 ERROR[2636]: chan_sip.c:11355 handle_request: Missing Cseq. Dropping this SIP message, it's incomplete. == Spawn extension (default, 200 at internal, 2) exited non-zero on 'Local/200 at inter...@default-f232,2' -- Executing DeadAGI(Local/200 at inter...@default-f232,2, agi:// 127.0.0.1:4577/call_log) in new stack == Spawn extension (from-ipkall, 901835, 3) exited non-zero on 'SIP/66.54.140.46-b781b470' -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing DeadAGI(Local/200 at inter...@default-f232,2, agi:// 127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-0-CANCEL--)) in new stack -- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-0-CANCEL--) completed, returning 0 == Refreshing DNS lookups. Any Solution ? ___ -- 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] Call Stealing
On Thursday, January 15, 2009, David fire wrote: hey it is preatty easy now i understand the problem is simple hangup in new location dial steal code for asterisk is just an extension and it should start an AGI the system search for the call in the same group bridge the channel to the current channel asterisk 1.6 or the system search for the call in the same group (AGI) send the channel to a conference (AGI search for the first free conference) join the current channel to the conference (AGI or AGI set a variable whit the conference number) That sounds like a reasonable idea. However, I've never written an AGI script and so I'm not sure how a script would detect which channel to steal. Checking through TFOT, I see there's CHANNEL STATUS - although I have no idea how to use it! Thanks for the pointer. -- Geoff ___ -- 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] How to transfer a call from one Asterisk Server to another
Why don't you simply Dial() the call to a separate box keeping Asterisk out of the audio path? l. 2009/1/16 Paul bulkm...@monafamily.com Can anyone tell me how I can completely move an established call off of one Asterisk server to another? In our case we have a server with our IVR. Depending upon digits entered, the call can be transferred to any of our other servers depending where the extension or queue reside. We would like to completely move the call off of the first box so we don't tie up resources on it. In our lab we are testing with 1.4.22.1 Our provider which delivers inbound calls to us uses a Sonus gateway. So far, testing has shown that if we transfer the inbound call prior to any media playback, it works. But, if the IVR plays media, then it is failing, with a 500 internal server error being returned. Thanks for any help ___ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- 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] Portech MV-378 with Asterisk
Emmanuel Pascal Bruno wrote: Has anyone been able to configure portech's mv-378 gateway with asterisk? I did the configuration as per the manual but it does not work. My server sees the portech gateway, but when the gateway is trying to register to my server it fails. It says peer is not suppose to register. The gateway and the asterisk box are on two different location (two network, 2 differrent IP address). I would appreciate any kind of tutorial or advice on how to make it work. Thanks ___ -- 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 Hi, I've an installation working with Portech MV-370. I'm supposing it's quite similar to what you have. If it could be useful to you, this is my sip.conf configuration file. [GSMGtw1] type=friend context=from-gsm host=dynamic; we have a DHCP assigned address secret=reallyverysecret nat=no ; there is not NAT between phone and Asterisk canreinvite=no dtmfmode=INFO insecure=invite ; required to overcome authentication problems in incoming calls call-limit=1 ; permit only 1 outgoing call at a time disallow=all allow=ulaw allow=alaw allow=gsm qualify=500 I remember that I've found a bug on the firmware that prevents to the unit to register correctly on my asterisk box unless I'm using the raw IP address instead of the name of the asterisk box. I remember something wrong in cryptography chiper/dechiper based on realm... So, if you have problems, let's try to specify the asterisk raw IP address in the Portech. Best regards, Marco Signorini. ___ -- 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] evaluate SIP response codes in dialplan
Johansson Olle E schrieb: 15 jan 2009 kl. 12.42 skrev Klaus Darilion: Johansson Olle E schrieb: 14 jan 2009 kl. 18.57 skrev Philipp Kempgen: Klaus Darilion schrieb: Philipp Kempgen schrieb: Klaus Darilion schrieb: Is it somehow possible to evaluate the SIP response code inside the dialplan? No. Part of the reasoning is that Asterisk is meant to be a multi- protocol PBX, not a SIP softswitch. This is IMO a stupid limitation. There are dozens of ISDN cause codes, dozens of SIP response codes and similar in other protocols, but Dial() only exports BUSY or CONGESTION .. I know. But the developers didn't want to add it. Which is incorrect. We don't want to add expose every protocol to the dialplan if not needed. As Josh and I've stated, we have the HANGUPCAUSE that gives you this level of detail, but in a multiprotocol way. The most important feature of Asterisk is that it's a multiprotocol PBX. Even if I think there's only one protocol for the future, there's still a lot of old stuff out there and the beauty is that I can produce services in asterisk covering all of these without knowing the details of all these protocols. It would be really bad if I had to write one app for every protocol covered by my dialplan. That's OK. HANGUPCAUSE is OK. Nevertheless a configurable mapping cause codes - SIP response codes would be nice :-) Absolutely - contact me off line to discuss such a project :-) In the meantime, we could document this a bit better. Yes - for example a note in the documentation of DIALSTATUS which refers to HANGUPCAUSE. One of the problems with hangupcause is, that is might get changed from one Asterisk to another - e.g. Hangup(3) generates a SIP 404 response which gets translated to hangupcause 1. So, a mechanism to signal Asterisk hangupcauses from one Asterisk to another Asterisk would be nice. IIRC I once saw a prorietary Asterisk header (X-Hangupcause or similar) in a SIP response, but I could find it currently. regards klaus regards klaus ___ -- 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] evaluate SIP response codes in dialplan
Are they identical to the ISDN hangup causes? http://networking.ringofsaturn.com/RemoteAccess/isdncausecodes.php klaus Johansson Olle E schrieb: 15 jan 2009 kl. 13.02 skrev John covici: That is very nice, but where are the HANGUPCAUSE values documented? That's the issue... include/asterisk/causes.h is a good reference for now. /O Thanks. on Thursday 01/15/2009 Johansson Olle E(o...@edvina.net) wrote 14 jan 2009 kl. 14.02 skrev Klaus Darilion: Hi! Is it somehow possible to evaluate the SIP response code inside the dialplan? I have an Asterisk server which forwards requests to various PSTN gateways with SIP. If the Dial() attempt is not successful I want to differ at least these 3 options: - called destination is busy (486): e.g. activate auto-redial - called destination does not exist, unassigned number (404) - gateway is broken, error, circuit busy (e.g. 503) 486 is mapped to DIALSTATUS=BUSY but both 503 and 404 is mapped to DIALSTATUS=CONGESTION As when Asterisk forwards the response with SIP to the caller the same response code is used, I suspect this information must be stored somewhere inside the channel variable. So, are there any means to access it? Check the HANGUPCAUSE, it's much more detailed than DIALSTATUS. We do map the SIP (and all other protocol errors in various channel drivers) codes to ISDN hangup causes, which gives you much more information about why a call failed. The conversion we're using follows the RFC, and where that doesn't cover it, Cisco's documentation. /Olle ___ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- 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 --- * Olle E Johansson - o...@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden ___ -- 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 ___ -- 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] multiple registration to sip trunking provider.
I do not know cordiip thus I do not know how these 3 different accounts are signaled to you, but some tips: A SIP peer is always identified by host:port - thus there is at peer level no way to differ them. But in the register command you specify the contact to be called, e.g. 1646H25. Thus, if you use 3 different contacts you should be able to differ the 3 accounts in the incoming context using 3 different dial patterns. regards klaus Andrea Borghi schrieb: a strange problem of multiple sip registrations and peer selection in sip.conf is calling for your suggestions!! let's examine this scenario: some numbers and passwords hidden with HHHs to protect the guilty :) I have 3 distinct sip subscriptions with cordiaip.net provider in US. For each of these i insert in sip.conf (with peer name differences and relefant number/password differences, of course] --- register = 1646H25:hh...@soft1.ny.cordiaip.net/1646H25 [cordiaus1] type=friend secret=H username=1646H25 fromuser=1646H25 fromdomain=soft1.ny.cordiaip.net host=soft1.ny.cordiaip.net call-limit=5 outboundproxy=soft1.ny.cordiaip.net disallow=all allow=gsm allow=alaw allow=ulaw context=DID_cordia insecure=port --- the sip registrations are OK and all seeems fine, BUT i have difficulties to map the incoming call because * is making mistakes in matching the incoming sip INVITE to the relevant peer. Please note that ALL the peers share the very same host and sip port. When i make a call to one of the subscribed cordia number, in sip debug i get a packes similar to this: --- SIP read from 38.98.115.34:5060 --- INVITE sip:16462487...@87.241.44.202 SIP/2.0 Via: SIP/2.0/UDP 38.98.115.34:5060;branch=z9hG4bK22981681-bdb335 To: sip:1646...@38.98.115.34 From: sip:39347...@38.98.115.34;tag=2298168-fdb335 Call-ID: 4926-0-1232058...@38.98.115.7 CSeq: 1 INVITE Contact: sip:393477135...@38.98.115.34:5060;transport=udp Server: Sansay-SIP/8.0 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 201 v=0 o=Sansay-SPX 11 11 IN IP4 38.98.115.9 s=Session Controller c=IN IP4 38.98.115.9 t=0 0 m=audio 15986 RTP/AVP 0 18 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 please note the From: and To: lines, I receive a From: with the caller CID (my mobile phone, in this case) and a To: with the sip number the call is directed to; this seems OK to me. I have read the chan_sip.c source file and it seems that when * receives this invite, it wals through the list of sip users/peers/friends to search for the correct entry from which cloning the sip parameters for the channel (as moh class, call limits, codecs and such) using the host IP as the key (if type=peer) or the caller number (if type=user) getting the values from the From: header. This seems very strange, because the user part of the From: header is potentially ANY number and the host part (and not the port, because is is always 5060 and there is insecure=port in place) in this scenario is not unique due to the 3 peers definitions. Please keep in mind that if i utilize only one registration i have absolutely no problems and can configure * correctly. The problem presents itself ONLY with multiple peers with multiple registrations to the same host/port. I cannot request cordia to forward me the numbers via an unique sip registration (sip trunking) because it seems that they don't offer this service. (but it may well be that i hadn't asked the right question) Can anyone suggest how to implement a correct sip trunking for this scenario, in which I have the incoming calls of the three registration going in a specific context (not the default, see context=DID_cordia in the peer definitions) and the outgoing calls going out via a specific user (so i can choose at the dialplan level with which number i am presenting myself in outgoing calls) I have spent some days trying various combinations of peers and users definitions, going in all cases to crash on the wall of the algorithm * uses to select the correct peer for the incoming calls. ___ -- 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 ___ -- 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] mISDN BRI Asterisk 1.4
Hi Francesco, You were correct. I pulled the cable out before everyone got in this morning and it was a cross over. I've now connected a proper straight-through ISDN cable (don't know what the Nortel was using before) and L1 is now up on Asterisk: BEGIN STACK_LIST: * Port 1 Type NT Prot. PTP L2Link DOWN L1Link:UP Blocked:0 Debug:1 However I'm now seeing the following message as L2 is not coming up: P[ 1] !!! Could not Get the L2 up after 3 Attemps!!! I checked that misdn.conf was setup to use PTMP (despite was it is saying above): mISDNconf module poll=128 debug=1 timer=nohfcmulti/module module debug=1 options=0mISDN_dsp/module devnode user=root group=root mode=644mISDN/devnode card type=BN4S0 port mode=nt link=ptmp1/port port mode=nt link=ptmp2/port port mode=nt link=ptmp3/port port mode=nt link=ptmp4/port /card /mISDNconf Also, how can I definately tell it is running in PTMP mode? msisdnportinfo does not distinguish: Port 1: NT-mode BRI S/T interface port (for phones) - Interface can be Poin-To-Point/Multipoint. Port 2: NT-mode BRI S/T interface port (for phones) - Interface can be Poin-To-Point/Multipoint. And Asterisk says it is running in PTP as well when the module is loaded: *CLI module load chan_misdn.so mISDN_close: fid(21) isize(131072) inbuf(0xb7b12008) irp(0xb7b12008) iend(0xb7b12008) == Parsing '/etc/asterisk/misdn.conf': Found P[ 0] Got: 1ptp,2ptp from get_ports == Registered channel type 'mISDN' (Channel driver for mISDN Support (Bri/Pri)) == Registered application 'misdn_set_opt' == Registered application 'misdn_facility' == Registered application 'misdn_check_l2l1' P[ 0] -- mISDN Channel Driver Registered -- Loaded chan_misdn.so = (Channel driver for mISDN Support (BRI/PRI)) I've got onto the Cisco router and found the BRI interface to be configured as below: interface BRI0 no ip address encapsulation ppp dialer pool-member 1 isdn switch-type basic-net3 no cdp enable ppp authentication chap callin end And the following events are being reported: Jan 16 10:03:06.099: ISDN BR0: Could not bring up interface Jan 16 10:03:06.103: BRI0: wait for isdn carrier timeout, call id=0x8004 The router also thinks that Layer 1 is still down: Global ISDN Switchtype = basic-net3 ISDN BRI0 interface dsl 0, interface ISDN Switchtype = basic-net3 Layer 1 Status: DEACTIVATED Layer 2 Status: Layer 2 NOT Activated Layer 3 Status: 0 Active Layer 3 Call(s) Active dsl 0 CCBs = 0 The Free Channel Mask: 0x8003 Looks like it may be worth speaking with OpenVox regarding exactly how there can is setup, only problem is I'm not sure what I should be asking them. Apologies for this long email. Regards Lee --- On Wed, 14/1/09, Francesco Peeters (linux) france...@fampeeters.com wrote: Are you using an ISDN cross cable? I don't know these cards, but most cards are wired as a DTE type device (TE port like a router or phone) and not a DCE type device (NT box). So you might have Tx-Tx and Rx-Tx instead of Rx-Tx and Tx-Rx... ;-) (Note that ISDN cross cables are definately NOT the same as a CAT5E cross cable!) ___ -- 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] Call Stealing
do you program in any language? if yes just read the chapters about agi in the asterisk book you can find it in support section in www.asterisk.org if you can't program send me an email I think this agi will be easy. I will program it for you (if you can't program) 2009/1/16, Geoff Lane ge...@gjctech.co.uk: On Thursday, January 15, 2009, David fire wrote: hey it is preatty easy now i understand the problem is simple hangup in new location dial steal code for asterisk is just an extension and it should start an AGI the system search for the call in the same group bridge the channel to the current channel asterisk 1.6 or the system search for the call in the same group (AGI) send the channel to a conference (AGI search for the first free conference) join the current channel to the conference (AGI or AGI set a variable whit the conference number) That sounds like a reasonable idea. However, I've never written an AGI script and so I'm not sure how a script would detect which channel to steal. Checking through TFOT, I see there's CHANNEL STATUS - although I have no idea how to use it! Thanks for the pointer. -- Geoff ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] want to add SipAddHeader in call out file
How to add SipAddHeader in outgoing call file. I am implementing a Callback scenario, in which a user makes a call to Local Access Number. The system have to callback to the user. During callback a call file is generated. All I want, is to add SipAddHeader(pchargingvector,val) in outgoing Invite. How can I achieve this? regards, Asif ___ -- 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] Call Stealing
On Friday, January 16, 2009, ddf...@gmail.com wrote: do you program in any language? if yes just read the chapters about agi in the asterisk book you can find it in support section in www.asterisk.org I'm a reasonable PHP and VBScript programmer and have dabbled since the 1980s in a wide range of languages from 6502 machine code upward. I've no Perl but I could learn or else use PHP. So, it would be an interesting exercise when I can find the time! Thanks again, -- Geoff ___ -- 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] want to add SipAddHeader in call out file
Use a Local/ channel in the Originate command, which can punt the outbound leg through dial plan logic that can call SipAddHeader() and tack on the header. Mian M Asif wrote: How to add SipAddHeader in outgoing call file. I am implementing a Callback scenario, in which a user makes a call to Local Access Number. The system have to callback to the user. During callback a call file is generated. All I want, is to add SipAddHeader(pchargingvector,val) in outgoing Invite. How can I achieve this? regards, Asif ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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] Call Stealing
2009/1/16 Geoff Lane ge...@gjctech.co.uk On Friday, January 16, 2009, ddf...@gmail.com wrote: do you program in any language? if yes just read the chapters about agi in the asterisk book you can find it in support section in www.asterisk.org I'm a reasonable PHP and VBScript programmer and have dabbled since the 1980s in a wide range of languages from 6502 machine code upward. I've no Perl but I could learn or else use PHP. So, it would be an interesting exercise when I can find the time! Thanks again, -- Geoff ___ -- 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 the book has a very good example in php. David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] Digium TE220 supported protocol
Laurent a écrit :
Those terms would be ISDN-related. VN4 is Version Number 4, and
ETSI is the European standards-adopting organization for telecoms.
So you might want to check for E1 support (ISDN in Europe,
basically) if you want to connect a PRI-capable equipment - I
assume that's what you are looking for since you mentioned the TE220.
If you read French, you might want to look at this page also:
http://blog.nicolargo.com/2008/01/installation-dune-carte-digium-avec-asterisk.html
I found it very useful when I installed recently a TE220 card.
Hi,
yes it's for connecting an PRI line. In fact i already have one PRI
connected to a 'real' PRI line,provided
by France Telecom. But this one is a little bit special, it's for
connecting to this kind of box:
http://www.oneaccess-net.com/en/one400.htm
Which is a SDSL/PRI gateway (yes, i asked for a direct SIP/H323
connection, but it's not in our product line was
the only answer ..).
Until know the L1 link doesn't goes up, so i'm looking for every bit of
info i can use. The FT line is configured like that:
span=1,1,0,ccs,hdb3
bchan=1-15
dchan=16
And work quite well, i've duplicated this for the access thru the One400
box:
span=2,0,0,ccs,hdb3
bchan=32-46
dchan=47
With no luck until now. I'm reading the one400 manual, and here are the
defaults setup for the PRI interface:
CLI(config)# interface pri 5/0
CLI(config-if)# physical-interface E1
CLI(config-if)# framing Df
CLI(config-if)# linecode hdb3
And here are the available framing:
CLI(config-e1)# framing {none | df | mf | emf}
Specifies the framing type.
None: no framing. Only used for CES / unstructured mode
Df: double frame, no CRC4. For E1 only. (Default value)
Mf: multiframe (CRC4). For E1 only
Emf: extended multiframe (CRC4). For E1 only.
Sf: Super-frame (for T1 only)
Esf: extended super frame (for T1 only)
I also found this:
The ONE 400 supports CCS mode (transport of
signaling messages over a D channel). CAS (Channel Associated
Signaling) mode is not supported.
However there is no references to this 'double frame' mode in the
dahdi/system.conf span configuration options
# framing::
# one of 'd4' or 'esf' for T1 or 'cas' or 'ccs' for E1 and BRI.
# 'd4' could be referred to as 'sf' or 'superframe'
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Re: [asterisk-users] Asterisk Upgrade
How can i erase asterisk 1.4 completely to reinstall 1.2.29 successfully again in steps please. From: torinti...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 16 Jan 2009 03:25:33 +0200 Subject: [asterisk-users] Asterisk Upgrade I was Using freepbx-2.1.1, Asterisk 1.2.29 and Asterisk Addons 1.2.9 i tried to upgrade to Asterisk 1.4.22.1 and Addons 1.4.7 all of the IAX trunks got not working at all. I tried to downgrade by make clean; make; make install in Atserisk 1.2.29 directory.but make gives errors in the end. How can i downgrade asterisk again and undo all changes i made?. (in steps please). and can Backup and Restore return all the previous asterisk configurations?. Thanks. Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! Try it! _ Show them the way! Add maps and directions to your party invites. http://www.microsoft.com/windows/windowslive/events.aspx___ -- 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] dialing trunk to trunk
Hello All,I'm very new in asterisk.Please help - how I can write conf files (or some example) for to delete one ext. and to add another, it means for example: I need to call from one asterisk to another by trunk to trunk and my dialing (for ex.) 100#...@1.2.1.2 when the the trunk of first asterisk is 1...@1.2.1.2 and trunk of second 1...@1.3.1.3, symbol '#' in this case doesn't important, I can use any another or some symbols. Sorry, if it is stupid, but I cannot get it. Thanks Leos -- We never did too much talking anyway So don't think twice, it's all right ___ -- 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] Asterisk Upgrade
1. rm -rf /var/lib/asterisk /var/spool/asterisk /etc/asterisk 2. Install 1.2.29. Torintino T wrote: How can i erase asterisk 1.4 completely to reinstall 1.2.29 successfully again in steps please. From: torinti...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 16 Jan 2009 03:25:33 +0200 Subject: [asterisk-users] Asterisk Upgrade I was Using freepbx-2.1.1, Asterisk 1.2.29 and Asterisk Addons 1.2.9 i tried to upgrade to Asterisk 1.4.22.1 and Addons 1.4.7 all of the IAX trunks got not working at all. I tried to downgrade by make clean; make; make install in Atserisk 1.2.29 directory.but make gives errors in the end. How can i downgrade asterisk again and undo all changes i made?. (in steps please). and can Backup and Restore return all the previous asterisk configurations?. Thanks. Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! Try it! http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us See all the ways you can stay connected to friends and family http://www.microsoft.com/windows/windowslive/default.aspx ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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] dialing trunk-to-trunk
Hello All,I'm very new in asterisk.Please help - how I can write conf files (or some example) for to delete one ext. and to add another, it means for example: I need to call from one asterisk to another by trunk to trunk and my dialing (for ex.) 100#...@1.2.1.2 when the the trunk of first asterisk is 1...@1.2.1.2 and trunk of second 1...@1.3.1.3, symbol '#' in this case doesn't important, I can use any another or some symbols. Sorry, if it is stupid, but I cannot get it. Thanks Leos -- We never did too much talking anyway So don't think twice, it's all right ___ -- 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] Digium TE220 supported protocol
Benoit a écrit :
Laurent a écrit :
Those terms would be ISDN-related. VN4 is Version Number 4, and
ETSI is the European standards-adopting organization for telecoms.
So you might want to check for E1 support (ISDN in Europe,
basically) if you want to connect a PRI-capable equipment - I
assume that's what you are looking for since you mentioned the TE220.
If you read French, you might want to look at this page also:
http://blog.nicolargo.com/2008/01/installation-dune-carte-digium-avec-asterisk.html
I found it very useful when I installed recently a TE220 card.
Hi,
yes it's for connecting an PRI line. In fact i already have one PRI
connected to a 'real' PRI line,provided
by France Telecom. But this one is a little bit special, it's for
connecting to this kind of box:
http://www.oneaccess-net.com/en/one400.htm
Which is a SDSL/PRI gateway (yes, i asked for a direct SIP/H323
connection, but it's not in our product line was
the only answer ..).
Until know the L1 link doesn't goes up, so i'm looking for every bit of
info i can use. The FT line is configured like that:
span=1,1,0,ccs,hdb3
bchan=1-15
dchan=16
And work quite well, i've duplicated this for the access thru the One400
box:
span=2,0,0,ccs,hdb3
bchan=32-46
dchan=47
With no luck until now. I'm reading the one400 manual, and here are the
defaults setup for the PRI interface:
CLI(config)# interface pri 5/0
CLI(config-if)# physical-interface E1
CLI(config-if)# framing Df
CLI(config-if)# linecode hdb3
And here are the available framing:
CLI(config-e1)# framing {none | df | mf | emf}
Specifies the framing type.
None: no framing. Only used for CES / unstructured mode
Df: double frame, no CRC4. For E1 only. (Default value)
Mf: multiframe (CRC4). For E1 only
Emf: extended multiframe (CRC4). For E1 only.
Sf: Super-frame (for T1 only)
Esf: extended super frame (for T1 only)
I also found this:
The ONE 400 supports CCS mode (transport of
signaling messages over a D channel). CAS (Channel Associated
Signaling) mode is not supported.
However there is no references to this 'double frame' mode in the
dahdi/system.conf span configuration options
# framing::
# one of 'd4' or 'esf' for T1 or 'cas' or 'ccs' for E1 and BRI.
# 'd4' could be referred to as 'sf' or 'superframe'
Also, i've used a straight Cat5E to connect the One400 to the TE220, both
seem to share the same Pin setup:
One 400:
1: RX (+)
2: RX (-)
4: TX (+)
5: TX (-)
TE 220:
1: RX
2: RX
4: TX
5: TX
Should i do some special crossing cable instead ? connecting Tx from one
side to other side Rx ?
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[asterisk-users] Can not fetch SIP_HEADER incase of Transfer
Hi, I am using Asterisk 1.4.12.1 version. Scenario for the call is as below: UAC Asterisk UAC Transfered to |--Invite-| | | | |---Invite| | | |--302 Moved-| | | |-Invite| I am setting a transfer context using set(__TRANSFER_CONTEXT=outgoing) . Now in outgoing context , I am fetching a SIP header. But asterisk gives me warnings as below do not returns value of sip header. WARNING[5410]: chan_sip.c:11678 function_sipchaninfo_read: This function can only be used on SIP channels. WARNING[5410]: chan_sip.c:11503 func_header_read: This function can only be used on SIP channels. Can you please let me know why this is happening? Why I do not get SIP header in transfer context? Is anybody there who have faced the same issue before? Thanks in advance!! -- Krunal Patel ___ -- 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] Asterisk Upgrade
Thanks. Date: Fri, 16 Jan 2009 07:15:29 -0500 From: abalas...@evaristesys.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk Upgrade 1. rm -rf /var/lib/asterisk /var/spool/asterisk /etc/asterisk 2. Install 1.2.29. Torintino T wrote: How can i erase asterisk 1.4 completely to reinstall 1.2.29 successfully again in steps please. From: torinti...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 16 Jan 2009 03:25:33 +0200 Subject: [asterisk-users] Asterisk Upgrade I was Using freepbx-2.1.1, Asterisk 1.2.29 and Asterisk Addons 1.2.9 i tried to upgrade to Asterisk 1.4.22.1 and Addons 1.4.7 all of the IAX trunks got not working at all. I tried to downgrade by make clean; make; make install in Atserisk 1.2.29 directory.but make gives errors in the end. How can i downgrade asterisk again and undo all changes i made?. (in steps please). and can Backup and Restore return all the previous asterisk configurations?. Thanks. Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! Try it! http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us See all the ways you can stay connected to friends and family http://www.microsoft.com/windows/windowslive/default.aspx ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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 _ Show them the way! Add maps and directions to your party invites. http://www.microsoft.com/windows/windowslive/events.aspx___ -- 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] Asterisk Upgrade
On Fri, 16 Jan 2009, Alex Balashov wrote: 1. rm -rf /var/lib/asterisk /var/spool/asterisk /etc/asterisk I'd suggest not removing /etc/asterisk if that's the only source of your config files... If you (re)generate them from elsewhere, it's probably OK. and the important one, I'd have thought is /usr/lib/asterisk/modules Gordon 2. Install 1.2.29. Torintino T wrote: How can i erase asterisk 1.4 completely to reinstall 1.2.29 successfully again in steps please. From: torinti...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 16 Jan 2009 03:25:33 +0200 Subject: [asterisk-users] Asterisk Upgrade I was Using freepbx-2.1.1, Asterisk 1.2.29 and Asterisk Addons 1.2.9 i tried to upgrade to Asterisk 1.4.22.1 and Addons 1.4.7 all of the IAX trunks got not working at all. I tried to downgrade by make clean; make; make install in Atserisk 1.2.29 directory.but make gives errors in the end. How can i downgrade asterisk again and undo all changes i made?. (in steps please). and can Backup and Restore return all the previous asterisk configurations?. Thanks. Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! Try it! http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us See all the ways you can stay connected to friends and family http://www.microsoft.com/windows/windowslive/default.aspx ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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 ___ -- 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] Asterisk Upgrade
Thanks to you. Date: Fri, 16 Jan 2009 13:24:16 + From: gordon+aster...@drogon.net To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk Upgrade On Fri, 16 Jan 2009, Alex Balashov wrote: 1. rm -rf /var/lib/asterisk /var/spool/asterisk /etc/asterisk I'd suggest not removing /etc/asterisk if that's the only source of your config files... If you (re)generate them from elsewhere, it's probably OK. and the important one, I'd have thought is /usr/lib/asterisk/modules Gordon 2. Install 1.2.29. Torintino T wrote: How can i erase asterisk 1.4 completely to reinstall 1.2.29 successfully again in steps please. From: torinti...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 16 Jan 2009 03:25:33 +0200 Subject: [asterisk-users] Asterisk Upgrade I was Using freepbx-2.1.1, Asterisk 1.2.29 and Asterisk Addons 1.2.9 i tried to upgrade to Asterisk 1.4.22.1 and Addons 1.4.7 all of the IAX trunks got not working at all. I tried to downgrade by make clean; make; make install in Atserisk 1.2.29 directory.but make gives errors in the end. How can i downgrade asterisk again and undo all changes i made?. (in steps please). and can Backup and Restore return all the previous asterisk configurations?. Thanks. Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! Try it! http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us See all the ways you can stay connected to friends and family http://www.microsoft.com/windows/windowslive/default.aspx ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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 ___ -- 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 _ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us___ -- 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] Portech MV-378 with Asterisk
Thanks for your reply! Can you tell me what you have in your Portech configuration settings (Mobile to Lan Settings; Sip Proxy settings etc...) My sip.conf file is pretty similar to yours but still cant register. On Fri, Jan 16, 2009 at 3:47 AM, Marco Signorini marcota...@libero.itwrote: Emmanuel Pascal Bruno wrote: Has anyone been able to configure portech's mv-378 gateway with asterisk? I did the configuration as per the manual but it does not work. My server sees the portech gateway, but when the gateway is trying to register to my server it fails. It says peer is not suppose to register. The gateway and the asterisk box are on two different location (two network, 2 differrent IP address). I would appreciate any kind of tutorial or advice on how to make it work. Thanks -- ___ -- 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 Hi, I've an installation working with Portech MV-370. I'm supposing it's quite similar to what you have. If it could be useful to you, this is my sip.conf configuration file. [GSMGtw1] type=friend context=from-gsm host=dynamic; we have a DHCP assigned address secret=reallyverysecret nat=no ; there is not NAT between phone and Asterisk canreinvite=no dtmfmode=INFO insecure=invite ; required to overcome authentication problems in incoming calls call-limit=1 ; permit only 1 outgoing call at a time disallow=all allow=ulaw allow=alaw allow=gsm qualify=500 I remember that I've found a bug on the firmware that prevents to the unit to register correctly on my asterisk box unless I'm using the raw IP address instead of the name of the asterisk box. I remember something wrong in cryptography chiper/dechiper based on realm... So, if you have problems, let's try to specify the asterisk raw IP address in the Portech. Best regards, Marco Signorini. ___ -- 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 ___ -- 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] want to add SipAddHeader in call out file
Dear Alex Balashov and All others,
can anyone give me the example how i can add local/channel with out
call file which used for Callback, Below is my dialplan for Callback.
Need to know where i can add SipAddHeader() in below dialplan. I want
to add in call leg one.
exten = _X.,1,wait(1)
exten = _X.,2,Set(outCallerID=${exten:1})
exten = _X.,3,Busy(1)
exten = _X.,4,Hangup()
exten = h,1,GotoIf($[${InvalidUser} = 1]?20:2)
exten = h,2,DeadAGI(STD/STD-CBLeg1-RadAuth.pl|${SIP_HEADER(Call-ID)})
exten = h,3,Set(CALLERID(number)=${CALLERID(number)})
exten = h,4,System(echo channel:
SIP/${callback...@${lcr_terminator_std} /tmp/${CALLERID(number)})
exten = h,5,System(echo context: STD-callback-leg2 /tmp/${CALLERID(number)})
exten = h,6,System(echo extension: s /tmp/${CALLERID(number)})
exten = h,7,System(echo priority: 1 /tmp/${CALLERID(number)})
exten = h,8,System(echo callerid: ${outCallerID}
/tmp/${CALLERID(number)}) ; Your CallerID goes here
exten = h,9,System(echo maxretries: 0 /tmp/${CALLERID(number)})
exten = h,10,System(echo retrytime: 3 /tmp/${CALLERID(number)})
exten = h,11,System(echo Set: confID=${confID} /tmp/${CALLERID(number)})
exten = h,12,System(echo Set: calltime=${calltime} /tmp/${CALLERID(number)})
exten = h,13,System(echo Set: CallBackNo=${CALLERID(number)}
/tmp/${CALLERID(number)})
exten = h,14,System(echo Set: Leg1CallID=${Leg1CallID}
/tmp/${CALLERID(number)})
exten = h,15,System(echo sleep 5 /tmp/${CALLERID(number)}.2)
exten = h,16,System(echo mv /tmp/${CALLERID(number)}
/var/spool/asterisk/outgoing /tmp/${CALLERID(number)}.2)
exten = h,17,System(chmod 775 /tmp/${CALLERID(number)}.2)
exten = h,18,System(/tmp/${CALLERID(number)}.2)
exten = h,19,NoOp(Hanging up ...!!)
exten = h,20,Hangup()
[STD-callback-leg2]
exten = s,1,NoOp(Entering callback-leg2)
exten = s,2,Set(CALLERID(number)=${CallBackNo})
;-- The Script Authorizes the user on Basis of Caller ID--
;-- Plays an IVR, gets destination Phno in SIP_Dest variable -
exten = s,3,Set(TIME_NOW=${EPOCH})
exten =
s,4,DeadAGI(STD/STD-CBLeg2-RadAuthAcc.pl|${confID}|${calltime}|${TIME_NOW}|${SIP_HEADER(Call-ID)}|${Leg1CallID})
exten = s,5,hangup()
Regards,
Asif
Date: Fri, 16 Jan 2009 06:17:35 -0500
From: Alex Balashov abalas...@evaristesys.com
Subject: Re: [asterisk-users] want to add SipAddHeader in call out
file
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: 49706ccf.8040...@evaristesys.com
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Use a Local/ channel in the Originate command, which can punt the
outbound leg through dial plan logic that can call SipAddHeader() and
tack on the header.
Mian M Asif wrote:
How to add SipAddHeader in outgoing call file.
I am implementing a Callback scenario, in which a user makes a call to
Local Access Number. The system have to callback to the user. During
callback a call file is generated. All I want, is to add
SipAddHeader(pchargingvector,val) in outgoing Invite.
How can I achieve this?
regards,
Asif
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Re: [asterisk-users] FWD and IPCall
David @ULC schrieb: Jan 15 17:18:43 ERROR[2636]: chan_sip.c:11355 handle_request: Missing Cseq. Dropping this SIP message, it's incomplete. This is the important part. One of your phones or FWD or IPCall does not speak SIP properly. Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] Signal Asterisk hangupcauses from one Asterisk to another (was: Re: evaluate SIP response codes in dialplan)
Klaus Darilion schrieb:
One of the problems with hangupcause is, that is might get changed from
one Asterisk to another - e.g. Hangup(3) generates a SIP 404 response
which gets translated to hangupcause 1. So, a mechanism to signal
Asterisk hangupcauses from one Asterisk to another Asterisk would be nice.
IIRC I once saw a prorietary Asterisk header (X-Hangupcause or similar)
in a SIP response, but I could find it currently.
asterisk 1:
SIPAddHeader(X-Hangupcause: 1);
Dial(SIP/${ext...@asterisk2);
asterisk 2:
if (${CHANNEL(channeltype)} = SIP) {
if (${SIP_HEADER(X-Hangupcause)} != ) {
Set(HANGUPCAUSE=${SIP_HEADER(X-Hangupcause)});
}
}
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
--
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[asterisk-users] pstn hangs up: MWI no message waiting ??
pstn incoming on a TDM400P, sometimes i* won't answer, going into a loop like this: -- Starting simple switch on 'DAHDI/4-1' [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 18 (Ring Begin)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 (Ring/Answered)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: Channel 4 no message waiting! -- Hungup 'DAHDI/4-1' I don't have any Message Waiting set ( or at least I don't think so.) Restarting * solves it for a while. Any suggestions? sean ___ -- 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] Signal Asterisk hangupcauses from one Asterisk to another
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
One of the problems with hangupcause is, that is might get changed from
one Asterisk to another - e.g. Hangup(3) generates a SIP 404 response
which gets translated to hangupcause 1. So, a mechanism to signal
Asterisk hangupcauses from one Asterisk to another Asterisk would be nice.
IIRC I once saw a prorietary Asterisk header (X-Hangupcause or similar)
in a SIP response, but I could find it currently.
asterisk 1:
SIPAddHeader(X-Hangupcause: 1);
Dial(SIP/${ext...@asterisk2);
asterisk 2:
if (${CHANNEL(channeltype)} = SIP) {
if (${SIP_HEADER(X-Hangupcause)} != ) {
Set(HANGUPCAUSE=${SIP_HEADER(X-Hangupcause)});
Don't know if that works. Maybe more like so:
Set(PRI_CAUSE=${SIP_HEADER(X-Hangupcause)});
Hangup(${SIP_HEADER(X-Hangupcause)});
}
}
Actually the other way round.
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
--
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Re: [asterisk-users] evaluate SIP response codes in dialplan
Klaus Darilion schrieb: Are they identical to the ISDN hangup causes? http://networking.ringofsaturn.com/RemoteAccess/isdncausecodes.php Yes. What you pass to Hangup() are Q.931 ISDN cause codes. See causes.h and hangup_cause2sip() in chan_sip.c for a list. Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] Snom 300 vs Grandstream gxp
Can anyone who has used both comment on the pros and cons ? Need to buy about 30 of these, for a small company with limited IT support. Julian __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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] Portech MV-378 with Asterisk
Pascal Bruno wrote: Thanks for your reply! Can you tell me what you have in your Portech configuration settings (Mobile to Lan Settings; Sip Proxy settings etc...) My sip.conf file is pretty similar to yours but still cant register. On Fri, Jan 16, 2009 at 3:47 AM, Marco Signorini marcota...@libero.it mailto:marcota...@libero.it wrote: Emmanuel Pascal Bruno wrote: Has anyone been able to configure portech's mv-378 gateway with asterisk? I did the configuration as per the manual but it does not work. My server sees the portech gateway, but when the gateway is trying to register to my server it fails. It says peer is not suppose to register. The gateway and the asterisk box are on two different location (two network, 2 differrent IP address). I would appreciate any kind of tutorial or advice on how to make it work. Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi, I've an installation working with Portech MV-370. I'm supposing it's quite similar to what you have. If it could be useful to you, this is my sip.conf configuration file. [GSMGtw1] type=friend context=from-gsm host=dynamic; we have a DHCP assigned address secret=reallyverysecret nat=no ; there is not NAT between phone and Asterisk canreinvite=no dtmfmode=INFO insecure=invite ; required to overcome authentication problems in incoming calls call-limit=1 ; permit only 1 outgoing call at a time disallow=all allow=ulaw allow=alaw allow=gsm qualify=500 I remember that I've found a bug on the firmware that prevents to the unit to register correctly on my asterisk box unless I'm using the raw IP address instead of the name of the asterisk box. I remember something wrong in cryptography chiper/dechiper based on realm... So, if you have problems, let's try to specify the asterisk raw IP address in the Portech. Best regards, Marco Signorini. Hi, I don't know if the problem could be in the Mobile to Lan or Lan to Mobile settings because these settings are related on how calls coming from/to mobile are routed. I didn't use the Portech routing features at all because I need a simple GSM gateway to/from the asterisk box. For this reason: 1. The only rule I've on Mobile to Lan is CID=*; url=...@192.168.0.5 where mob is the extension I've generated in the asterisk box under the context where the Portech operates; 2. The only rule I've on Lan to Mobile is URL=*; Call Num=# I think the most relevant parameters for your problem are under the Service Domain menu option (assuming that the firmware you have is similar to what I've). On this menu I've compiled the 1st Realm (as I've only one account) like that: UserName: GSMGtw1 RegisterName: GSMGtw1 RegisterPassword: reallyverysecret Domain Server: 192.168.0.5 Proxy Server: 192.168.0.5 Pay attention that, having specified the Domain Server with the raw IP address, asterisk needs to be able to authenticate peers associated to that. For this reason I've set: domain=192.168.0.5 on sip.conf [general] section (remember to issue a sip reload from asterisk cli). Hope this helps! Best regards. Marco Signorini Marco Signorini INGEGNI Tech S.r.l. http://www.ingegnitech.com ___ -- 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] Snom 300 vs Grandstream gxp
Hi, I've never used Snom phones, but have used the Grandstreams. I think you will find that they just feel 'cheap.' We had a half dozen of them, and the functionality is there, and they work great. But they just feel rough and cheap when using them. If you are planning on using different headsets with them, you are fine. But if you are planning on using the factory headsets, you might find that the headset has rough edges, etc. Call me 'crazy'! We're using Linksys SPA-942/941's and couldn't be happier. The 941 model is a dollar or so more than the GXP, but don't have dual Ethernet. 942's do, for an extra $20. Regards, Robert Broyles Julian Lyndon-Smith wrote: Can anyone who has used both comment on the pros and cons ? Need to buy about 30 of these, for a small company with limited IT support. Julian __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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 ___ -- 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] Snom 300 vs Grandstream gxp
On Fri, 16 Jan 2009, Julian Lyndon-Smith wrote: Can anyone who has used both comment on the pros and cons ? Need to buy about 30 of these, for a small company with limited IT support. You get more phone for your money with the Grandstream, but ... You'll hear a lot of people here who've had bad results with them in the past, so will be baised against them, and there were early issues with software quality - and still some issues with newer software on holder handsets, however ... I've deployed a lot of Grandstreams - 100's. Mostly GXP2000s but one place (a small school on a tight budget) went for BT200's with a single GXP2000 for the reception and they're happy. If budget was that tight, I'd look at the GXP280's though. Audio quality in the Snoms is probably better. Build quality is a little better too - the handsets are slightly heavier. I find call transfers easier on the Grandstreams - also the display (on the GXP range) is much bigger than that Snom 300's. That's nice if you're putting up number-names. The web interface on both is easy enough if you're not auto provisioning. (I don't auto provision the grandstreams as such, but use a utility called 'gsutil' on them) However, while it's very comprehensive, I think there's too much information there! Well - that's my input... (mostly on Grandstreams though!) Gordon ___ -- 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] Remote RTP
Hi all, Suposing that 2 SIP phone register at a remote (internet) asterisk, what is the best way, if any, to make the RTP traffic go phone to phone, whithout using the internet conection (asterisk)? Thanks, Gabriel ___ -- 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] Remote RTP
canreinvite=yes. Gabriel Ortiz Lour wrote: Hi all, Suposing that 2 SIP phone register at a remote (internet) asterisk, what is the best way, if any, to make the RTP traffic go phone to phone, whithout using the internet conection (asterisk)? Thanks, Gabriel ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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] Remote RTP
On Jan 16, 2009, at 10:38 AM, Gabriel Ortiz Lour wrote: Hi all, Suposing that 2 SIP phone register at a remote (internet) asterisk, what is the best way, if any, to make the RTP traffic go phone to phone, whithout using the internet conection (asterisk)? Allow reinvite? Assuming both are not behind NAT. ___ -- 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] Remote RTP
They will be in the same LAN, probably behind NAT. Being in the same LAN helps something? 2009/1/16 Jerry Jones jjo...@danrj.com On Jan 16, 2009, at 10:38 AM, Gabriel Ortiz Lour wrote: Hi all, Suposing that 2 SIP phone register at a remote (internet) asterisk, what is the best way, if any, to make the RTP traffic go phone to phone, whithout using the internet conection (asterisk)?They Allow reinvite? Assuming both are not behind NAT. ___ -- 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 ___ -- 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] Remote RTP
Gabriel Ortiz Lour wrote: Hi all, Suposing that 2 SIP phone register at a remote (internet) asterisk, what is the best way, if any, to make the RTP traffic go phone to phone, whithout using the internet conection (asterisk)? Thanks, Gabriel By default, Asterisk will attempt to offload the media from the server so that it may flow directly between the phones. There are several factors which may prevent this, though. For instance, if Asterisk is recording the call or needs to listen for DTMF in order to activate a specific feature, then Asterisk has to have the RTP flow through it. Mark Michelson ___ -- 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] Portech MV-378 with Asterisk
Thank you!, I will try that in a few hours and let you know what happens. On Fri, Jan 16, 2009 at 11:01 AM, Marco Signorini marcota...@libero.itwrote: Pascal Bruno wrote: Thanks for your reply! Can you tell me what you have in your Portech configuration settings (Mobile to Lan Settings; Sip Proxy settings etc...) My sip.conf file is pretty similar to yours but still cant register. On Fri, Jan 16, 2009 at 3:47 AM, Marco Signorini marcota...@libero.itwrote: Emmanuel Pascal Bruno wrote: Has anyone been able to configure portech's mv-378 gateway with asterisk? I did the configuration as per the manual but it does not work. My server sees the portech gateway, but when the gateway is trying to register to my server it fails. It says peer is not suppose to register. The gateway and the asterisk box are on two different location (two network, 2 differrent IP address). I would appreciate any kind of tutorial or advice on how to make it work. Thanks -- ___ -- 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 Hi, I've an installation working with Portech MV-370. I'm supposing it's quite similar to what you have. If it could be useful to you, this is my sip.conf configuration file. [GSMGtw1] type=friend context=from-gsm host=dynamic; we have a DHCP assigned address secret=reallyverysecret nat=no ; there is not NAT between phone and Asterisk canreinvite=no dtmfmode=INFO insecure=invite ; required to overcome authentication problems in incoming calls call-limit=1 ; permit only 1 outgoing call at a time disallow=all allow=ulaw allow=alaw allow=gsm qualify=500 I remember that I've found a bug on the firmware that prevents to the unit to register correctly on my asterisk box unless I'm using the raw IP address instead of the name of the asterisk box. I remember something wrong in cryptography chiper/dechiper based on realm... So, if you have problems, let's try to specify the asterisk raw IP address in the Portech. Best regards, Marco Signorini. Hi, I don't know if the problem could be in the Mobile to Lan or Lan to Mobile settings because these settings are related on how calls coming from/to mobile are routed. I didn't use the Portech routing features at all because I need a simple GSM gateway to/from the asterisk box. For this reason: 1. The only rule I've on Mobile to Lan is CID=*; url=...@192.168.0.5 where mob is the extension I've generated in the asterisk box under the context where the Portech operates; 2. The only rule I've on Lan to Mobile is URL=*; Call Num=# I think the most relevant parameters for your problem are under the Service Domain menu option (assuming that the firmware you have is similar to what I've). On this menu I've compiled the 1st Realm (as I've only one account) like that: UserName: GSMGtw1 RegisterName: GSMGtw1 RegisterPassword: reallyverysecret Domain Server: 192.168.0.5 Proxy Server: 192.168.0.5 Pay attention that, having specified the Domain Server with the raw IP address, asterisk needs to be able to authenticate peers associated to that. For this reason I've set: domain=192.168.0.5 on sip.conf [general] section (remember to issue a sip reload from asterisk cli). Hope this helps! Best regards. Marco Signorini Marco Signorini INGEGNI Tech S.r.l. http://www.ingegnitech.com ___ -- 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 ___ -- 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] Snom 300 vs Grandstream gxp
On Jan 16, 2009, at 7:52 AM, Julian Lyndon-Smith wrote: Can anyone who has used both comment on the pros and cons ? Need to buy about 30 of these, for a small company with limited IT support. We recently deployed 85 phones to our office. We tested the Grandstream GXP2000, GXP2020, Linksys SPA941, Snom 300 320, and a Polycom 430 (I think that was the series). As an IT department we expected everybody to prefer the Grandstream because it is simple to use. We figured everybody would have the Snom because it is complex to use (though super easy on the IT side to administer). We had the opposite result. Everybody hated the Grandstream because they sounded bad, felt clunky, were difficult to do simple things on (like park a call, can't do it with one button). Nobody really cared for either the Linksys or Polycom. They were just too limited. We ended up rolling out a mixture of the Snom 300 and 320s and couldn't be happier (We looked at the 360, but it really doesn't offer anything except a bigger display, which isn't really utilized). With a simple MySQL database and a few PHP scripts all we had to do was type the MAC address of the phone into the MySQL database (with the login information) and then plug the phones in. No setup on the phone. Phone automatically upgrades the firmware to whatever version we currently use, gets its settings from the server, etc. If a phone has trouble (out of the 85 we had 2 that were a bit finicky and got replaced), we go into the database and change the MAC address and then plug in the new phone. Again, no setup. If you go Snom I would be happy to share these scripts, I just haven't gotten around to building up a nice package and posting them. If your choices are either Snom or Grandstream, I would so go Snom. I spent 2 days trying to configure the GXP's to do the few simple things we wanted and couldn't pull it off (call parking, BLF one-touch dial [does not fully work], etc). I spent 30 minutes on the Snom an had it perfectly configured. Julian Daniel ___ -- 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] Dialing from E1/T1
Hi, A have an asterisk connected to a legacy PBX trought an E1 and to the PSTN trought another E1. When the legacy user dial to the PSTN the call pass trought Asterisk. All works OK, the only problem is the delay on the Asterisk server when it receives the digits from the 1st E1 link. It will only make the call when the digit timeout expires. Is there a way to make something like dialplan/context/exten match, so it will dial as soon as there a match? Thanks Gabriel Ortiz ___ -- 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] How to transfer a call from one AsteriskServer to another
I do have it functioning with Dial(). I was looking for a way to completely move the call from the first box though. When using Dial() media moves, but the call is still tied to the first box. In looking at captures when the call is ended, the first box invites out to the ITSP again, then after receiving a 200ok sends a bye. Also while testing, once the call was up on the second box, I stopped Asterisk on the first box which kills the call. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri Sent: Friday, January 16, 2009 12:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to transfer a call from one AsteriskServer to another Why don't you simply Dial() the call to a separate box keeping Asterisk out of the audio path? l. 2009/1/16 Paul bulkm...@monafamily.com Can anyone tell me how I can completely move an established call off of one Asterisk server to another? In our case we have a server with our IVR. Depending upon digits entered, the call can be transferred to any of our other servers depending where the extension or queue reside. We would like to completely move the call off of the first box so we don't tie up resources on it. In our lab we are testing with 1.4.22.1 Our provider which delivers inbound calls to us uses a Sonus gateway. So far, testing has shown that if we transfer the inbound call prior to any media playback, it works. But, if the IVR plays media, then it is failing, with a 500 internal server error being returned. Thanks for any help ___ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- 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] How to hangup a call manually...
I have this call: SIP/protel-525512047 default 90445528885371 1 Ringing AppDial (Outgoing Line) 90445528885371 264:24:2 (None) I cannot use the soft hangup commando from the CLI because I do not know the whole SIP channel string. What other command can I use to terminate this call or to find the complete channel string to put into soft hangup? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- 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] How to hangup a call manually...
try to know the whole string ? core show channels 2009/1/16 Carlos Chavez cur...@telecomabmex.com I have this call: SIP/protel-525512047 default 90445528885371 1 Ringing AppDial (Outgoing Line) 90445528885371 264:24:2 (None) I cannot use the soft hangup commando from the CLI because I do not know the whole SIP channel string. What other command can I use to terminate this call or to find the complete channel string to put into soft hangup? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- 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 ___ -- 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] How to hangup a call manually...
2009/1/17 Carlos Chavez cur...@telecomabmex.com I have this call: SIP/protel-525512047 default 90445528885371 1 Ringing AppDial (Outgoing Line) 90445528885371 264:24:2 (None) I cannot use the soft hangup commando from the CLI because I do not know the whole SIP channel string. What other command can I use to terminate this call or to find the complete channel string to put into soft hangup? It looks like you have the entire channel string there, but, if not... The CLI support tab completion, so soft hangup SIP/protel-tab will either fully autocomplete, or, where there are multiple matches, give you a list of matching channels... d ___ -- 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] How to hangup a call manually...
If youre using the GUI it will hang it up. Otherwise sip reload might do it. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Grygoriy Dobrovolskyy Sent: Friday, January 16, 2009 12:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to hangup a call manually... try to know the whole string ? core show channels 2009/1/16 Carlos Chavez cur...@telecomabmex.com I have this call: SIP/protel-525512047 default 90445528885371 1 Ringing AppDial (Outgoing Line) 90445528885371 264:24:2 (None) I cannot use the soft hangup commando from the CLI because I do not know the whole SIP channel string. What other command can I use to terminate this call or to find the complete channel string to put into soft hangup? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- 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 ___ -- 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] Voicemail message is dialtone
Hello all, I have one Asterisk 1.4.21 system connected to a North American POTS line. Normally hangup detection works fine, and Asterisk hangs up properly if you are talking to a caller and they hang up; but occasionally a call comes in (typically from a US telemarketer) where the caller hangs up just as voicemail recording is starting, and you get a voicemail recording of dialtone (then congestion and off-hook warning tones) for almost 4 minutes before asterisk gives up the line. zapata.conf [channels] options are: language=en context=default rxwink=300 usecallerid=yes hidecallerid=no cidsignalling=bell callwaiting=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=no transfer=yes canpark=yes cancallforward=no callreturn=no musiconhold=default echocancel=yes echocancelwhenbridged=yes immediate=no faxdetect=no relaxdtmf=yes hanguponpolarityswitch=yes progzone=us signalling=fxs_ks channel = 1 Any suggestions for voicemail detecting/rejecting messages when there is only dialtone on the other end? Thanks! S. ___ -- 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] How to transfer a call from one AsteriskServer to another
I guess you already tried this? http://www.voip-info.org/wiki-Asterisk+cmd+Transfer Thanks l. 2009/1/16 Paul bulkm...@monafamily.com I do have it functioning with Dial(). I was looking for a way to completely move the call from the first box though. When using Dial() media moves, but the call is still tied to the first box. In looking at captures when the call is ended, the first box invites out to the ITSP again, then after receiving a 200ok sends a bye. Also while testing, once the call was up on the second box, I stopped Asterisk on the first box which kills the call. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Lenz Emilitri *Sent:* Friday, January 16, 2009 12:17 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] How to transfer a call from one AsteriskServer to another Why don't you simply Dial() the call to a separate box keeping Asterisk out of the audio path? l. 2009/1/16 Paul bulkm...@monafamily.com Can anyone tell me how I can completely move an established call off of one Asterisk server to another? In our case we have a server with our IVR. Depending upon digits entered, the call can be transferred to any of our other servers depending where the extension or queue reside. We would like to completely move the call off of the first box so we don't tie up resources on it. In our lab we are testing with 1.4.22.1 Our provider which delivers inbound calls to us uses a Sonus gateway. So far, testing has shown that if we transfer the inbound call prior to any media playback, it works. But, if the IVR plays media, then it is failing, with a 500 internal server error being returned. Thanks for any help ___ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- 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] Dialing from E1/T1
exten = _X,1,Dial(DAHDI/g1/${EXTEN})
or
exten = _9,1,Dial(DAHDI/g1/${EXTEN:1})
the first whan put X to match the amount of digits.
the second one dial if you put a nine before. and many X as digits
David
2009/1/16 Gabriel Ortiz Lour ortiz.ad...@gmail.com
Hi,
A have an asterisk connected to a legacy PBX trought an E1 and to the
PSTN trought another E1. When the legacy user dial to the PSTN the call pass
trought Asterisk.
All works OK, the only problem is the delay on the Asterisk server when
it receives the digits from the 1st E1 link. It will only make the call when
the digit timeout expires.
Is there a way to make something like dialplan/context/exten match, so it
will dial as soon as there a match?
Thanks
Gabriel Ortiz
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[asterisk-users] CDR problems -- two call legs create only one CDR. Using ForkCDR() not even working.
Hello, When I bridge an incoming and outgoing call (attempting to simulate call-forwarding) I'm only getting one CDR -- that of the outgoing call. A (PSTN) calls B (residing on Asterisk) and the Asterisk calls C (cell phone on PSTN) and bridges the call. The only CDR created is from B to C. I have even tried using Answer() and ForkCDR() to get two CDRs, but to no avail. I am starting to wonder if there's a bug in the CDR generation in general, because I set up an extension to do only that: exten = 5822558,1,Answer() exten = 5822558,n,ForkCDR() exten = 5822558,n,Playback(tt-monkeys) exten = 5822558,n,Hangup() This is even given as an example on how to generate two CDRs from one call on this website: http://asterisk.name/asterisk/0596009623/asterisk-app-b-79.html I have been able to create two CDRs with the use of the Local/n channel, but the CDR is messy if I do so because I am required by law to change the caller-id for the outgoing call to that of the PBX, so both call legs seem to be originating from the Asterisk. Am I missing something? Any ideas appreciated. Best regards, Örn ___ -- 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] Voicemail message is dialtone
Here's also an example snip from the debug log: [07:42:20] -- Executing [...@mainmenu:15] Dial(Zap/1-1, SIP/105|18|tKk) in new stack [07:42:20] -- SIP/105-08571180 is ringing [07:42:39] -- Nobody picked up in 18000 ms [07:42:39] -- Executing [...@mainmenu:16] Answer(Zap/1-1, ) in new stack [07:42:39] -- Executing [...@mainmenu:17] Wait(Zap/1-1, 1) in new stack [07:42:40] -- Executing [...@mainmenu:18] Playback(Zap/1-1, silence/1) in new stack [07:42:40] -- Zap/1-1 Playing 'silence/1' (language 'en') [07:42:41] -- Executing [...@mainmenu:19] BackGround(Zap/1-1, please-leave-a-message) in new stack [07:42:41] -- Zap/1-1 Playing 'please-leave-a-message' (language 'en') [07:42:47] -- Executing [...@mainmenu:20] VoiceMail(Zap/1-1, 105|s) in new stack [07:42:48] -- Zap/1-1 Playing 'beep' (language 'en') [07:42:48] -- Recording the message [07:42:48] -- x=0, open writing: /var/spool/asterisk/voicemail/default/100/tmp/UNCGT6 format: wav49, 0x8570eb8 [07:42:48] -- x=1, open writing: /var/spool/asterisk/voicemail/default/100/tmp/UNCGT6 format: gsm, 0x8227430 [07:42:48] -- x=2, open writing: /var/spool/asterisk/voicemail/default/100/tmp/UNCGT6 format: wav, 0x8259b10 [07:46:42] -- Recording automatically stopped after a silence of 10 seconds [07:46:42] -- Zap/1-1 Playing 'auth-thankyou' (language 'en') [07:46:43] == Auto fallthrough, channel 'Zap/1-1' status is 'NOANSWER' [07:46:43] -- Hungup 'Zap/1-1' On Fri, Jan 16, 2009 at 12:07 PM, Steve Johnson stevej...@gmail.com wrote: Hello all, I have one Asterisk 1.4.21 system connected to a North American POTS line. Normally hangup detection works fine, and Asterisk hangs up properly if you are talking to a caller and they hang up; but occasionally a call comes in (typically from a US telemarketer) where the caller hangs up just as voicemail recording is starting, and you get a voicemail recording of dialtone (then congestion and off-hook warning tones) for almost 4 minutes before asterisk gives up the line. zapata.conf [channels] options are: language=en context=default rxwink=300 usecallerid=yes hidecallerid=no cidsignalling=bell callwaiting=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=no transfer=yes canpark=yes cancallforward=no callreturn=no musiconhold=default echocancel=yes echocancelwhenbridged=yes immediate=no faxdetect=no relaxdtmf=yes hanguponpolarityswitch=yes progzone=us signalling=fxs_ks channel = 1 Any suggestions for voicemail detecting/rejecting messages when there is only dialtone on the other end? Thanks! S. ___ -- 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] How to transfer a call from one AsteriskServerto another
Yes, this is the first method I tried. The transfer only works if it is done before a media path is set up to the first box (not answered by the IVR). If it is answered then transferred, I get a 500 internal server error back from the ITSP and the call dies. I never see anything hit the second box. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri Sent: Friday, January 16, 2009 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to transfer a call from one AsteriskServerto another I guess you already tried this? http://www.voip-info.org/wiki-Asterisk+cmd+Transfer Thanks l. 2009/1/16 Paul bulkm...@monafamily.com I do have it functioning with Dial(). I was looking for a way to completely move the call from the first box though. When using Dial() media moves, but the call is still tied to the first box. In looking at captures when the call is ended, the first box invites out to the ITSP again, then after receiving a 200ok sends a bye. Also while testing, once the call was up on the second box, I stopped Asterisk on the first box which kills the call. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri Sent: Friday, January 16, 2009 12:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to transfer a call from one AsteriskServer to another Why don't you simply Dial() the call to a separate box keeping Asterisk out of the audio path? l. 2009/1/16 Paul bulkm...@monafamily.com Can anyone tell me how I can completely move an established call off of one Asterisk server to another? In our case we have a server with our IVR. Depending upon digits entered, the call can be transferred to any of our other servers depending where the extension or queue reside. We would like to completely move the call off of the first box so we don't tie up resources on it. In our lab we are testing with 1.4.22.1 Our provider which delivers inbound calls to us uses a Sonus gateway. So far, testing has shown that if we transfer the inbound call prior to any media playback, it works. But, if the IVR plays media, then it is failing, with a 500 internal server error being returned. Thanks for any help ___ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- 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] pstn hangs up: MWI no message waiting ??
- sean darcy seandar...@gmail.com wrote: pstn incoming on a TDM400P, sometimes i* won't answer, going into a loop like this: -- Starting simple switch on 'DAHDI/4-1' [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 18 (Ring Begin)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 (Ring/Answered)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: Channel 4 no message waiting! This message occurs when the pstn sends a FSK spill indicating the message waiting status of the FXO port in question. This may encoded in the caller ID indicator or may be contained in its own message spill. This is output as a NOTICE logging message. Regards, Doug Bailey ___ -- 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] mini-PCI FXS card?
Is there any product that's a single port mini-PCI FXS card? I'm aware of the Openvox A400M http://www.openvox.com.cn/products.php?genre_id=39, but I really only wanted one port. How about a single or dual port PCI or PCI express FXS card? Basically I wanted to build a small linux router with one or two phone ports. Alternatively, is there already a router or single board computer with FXS ports that I could run linux/asterisk on? ___ -- 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] mini-PCI FXS card?
Adam Moffett schrieb: How about a single or dual port PCI or PCI express FXS card? Basically I wanted to build a small linux router with one or two phone ports. I'd recommend Sangoma's new B700 FlexBRI hybrid card (4 BRI ports, 2 FXS/FXO) http://www.sangoma.com/products_and_solutions/hardware/digital_analog_hybrids/flex_bri.html or the B600 (4 FXO, 1 FXS) http://www.sangoma.com/products_and_solutions/hardware/analog_telephony/b600.html although that might be a bit mor than you need. Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] mini-PCI FXS card?
Thanks Philipp, This and everything else I see out there is a bit more than I need :) I'm sure a single or dual port analog FXS card is not something most people want though, otherwise somebody would be selling it. Thanks anyway though. I'd recommend Sangoma's new B700 FlexBRI hybrid card (4 BRI ports, 2 FXS/FXO) http://www.sangoma.com/products_and_solutions/hardware/digital_analog_hybrids/flex_bri.html or the B600 (4 FXO, 1 FXS) http://www.sangoma.com/products_and_solutions/hardware/analog_telephony/b600.html although that might be a bit mor than you need. Philipp Kempgen ___ -- 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] about hardware
hi i want to setup an asterisk in my home i always worked whit digium hardware but have any one try the chines cards or the openbox or sangoma? is just for home 2 FXS and 2 FXO i need an aceptable audio and the pc will do only asterisk and is a dual core 1G RAM the cheaper chines cards will be ok? thanks -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] ATA gateway with 2 ethernet interfaces
Hi, I'm looking for an 8+ FXS ATA gateway (at least 8 ports but preferably at most 24 ports) with 2 ethernet interfaces for network/switch redundancy. So far I've only found the Grandstream GXW4008. I've searched similar brands such as Linksys and higher-end brands such as Quintum, but they all seem to have just one NIC. So, if the switch the ATA is connected to fails then I'm out of business (at least until I replace the switch but that's usually too long for a busy system). The GXW4008 device is very useful for this scenario. It has 2 RJ45 ports (called WAN and LAN) and I've set them up in two local subnets. Not only does the ATA keep working without human intervention if one of the switches goes down but if both switches are up it can load balance between the two (simply by using DNS SRV with the same weights). Unfortunately, Grandstream in general doesn't seem to be very reliable although the latest GXW4008 firmware has proven to be quite stable in my case (previous releases were buggy). So I'm looking for alternatives to the GXW4008, even if it has to cost me more money. Does anyone know of an 8+ FXS ATA brand/model with 2 ethernet interfaces? Thanks in advance. Vieri ___ -- 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] Crickets. Yes, crickets.
How many times have you been on a conference call and some other participant puts their line on hold, leading to their hold music making conversation impossible for the rest of the group? This scenario happens to me all the time and it drives me NUTS. I prefer no hold music, because I am often on conference calls, and I also usually hate listening to music on the phone when I usually have my own music playing in the background as well. And most hold music is really bad. However, there is a valid reason for hold music and that is to let the caller know via an audio cue that they have not been disconnected. So I heard about some really great hold audio the other day. It was the sound of crickets chirping quietly, every 5-8 seconds, seemingly randomly. It was actually just a 2 or 3 minute recording of a few different cricket sounds, recorded at low gain. It was unobtrusive enough to not interrupt speakers on the call, yet if it was for a single listener it was enough audio to be obvious that the other speaker was still on hold. I have made some of the sounds in the past (tt-monkeys.gsm comes to mind) but at the moment I am pretty much tapped out for time and even trivial things like making sound files are difficult, though I can type messages to asterisk-users well enough, as I sit ironically on hold. Does anyone want to take up the task? While as usual I can't guarantee inclusion of any resulting soundfile into Asterisk, it would certainly get _my_ vote. JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- 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] Asterisk 1.4.23-rc4 Now Available
The Asterisk.org development team has published Asterisk 1.4.23-rc4. This release candidate is available for download from http://downloads.digium.com/. A number of critical issues have been resolved since the last release candidate for 1.4.23. We hope to have this be the final release candidate. If all goes well, 1.4.23 will be released early next week. For a full list of changes, see the ChangeLog: http://svn.digium.com/svn-view/asterisk/tags/1.4.23-rc4/ChangeLog?view=markup Testing of this release candidate would be very much appreciated. One area that has received a lot of changes is the handling of call parking, so that is an area that could use some special attention in testing. Please report any issues found in testing to http://bugs.digium.com/. Thank you for your support of Asterisk! ___ -- 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] ATA gateway with 2 ethernet interfaces
I don't know of any ATA like that except the grandstream. The service provider grade way to do this would probably be a Cisco (or similar) with a T1 interface and a channel bank to break the T1 into 24 FXS ports. Hi, I'm looking for an 8+ FXS ATA gateway (at least 8 ports but preferably at most 24 ports) with 2 ethernet interfaces for network/switch redundancy. So far I've only found the Grandstream GXW4008. I've searched similar brands such as Linksys and higher-end brands such as Quintum, but they all seem to have just one NIC. So, if the switch the ATA is connected to fails then I'm out of business (at least until I replace the switch but that's usually too long for a busy system). The GXW4008 device is very useful for this scenario. It has 2 RJ45 ports (called WAN and LAN) and I've set them up in two local subnets. Not only does the ATA keep working without human intervention if one of the switches goes down but if both switches are up it can load balance between the two (simply by using DNS SRV with the same weights). Unfortunately, Grandstream in general doesn't seem to be very reliable although the latest GXW4008 firmware has proven to be quite stable in my case (previous releases were buggy). So I'm looking for alternatives to the GXW4008, even if it has to cost me more money. Does anyone know of an 8+ FXS ATA brand/model with 2 ethernet interfaces? Thanks in advance. Vieri ___ -- 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 ___ -- 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] ATA gateway with 2 ethernet interfaces
I would be more worried about the ATA gateway failing than the switch, as you have found yourself. How about two gateways and two phones on everyone's desk :) j On Fri, 16 Jan 2009, Adam Moffett wrote: I don't know of any ATA like that except the grandstream. The service provider grade way to do this would probably be a Cisco (or similar) with a T1 interface and a channel bank to break the T1 into 24 FXS ports. Hi, I'm looking for an 8+ FXS ATA gateway (at least 8 ports but preferably at most 24 ports) with 2 ethernet interfaces for network/switch redundancy. So far I've only found the Grandstream GXW4008. I've searched similar brands such as Linksys and higher-end brands such as Quintum, but they all seem to have just one NIC. So, if the switch the ATA is connected to fails then I'm out of business (at least until I replace the switch but that's usually too long for a busy system). The GXW4008 device is very useful for this scenario. It has 2 RJ45 ports (called WAN and LAN) and I've set them up in two local subnets. Not only does the ATA keep working without human intervention if one of the switches goes down but if both switches are up it can load balance between the two (simply by using DNS SRV with the same weights). Unfortunately, Grandstream in general doesn't seem to be very reliable although the latest GXW4008 firmware has proven to be quite stable in my case (previous releases were buggy). So I'm looking for alternatives to the GXW4008, even if it has to cost me more money. Does anyone know of an 8+ FXS ATA brand/model with 2 ethernet interfaces? Thanks in advance. Vieri ___ -- 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 ___ -- 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 ___ -- 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] Asterisk - Trixbox
On Thu, Jan 15, 2009 at 8:00 PM, Mike Hammett asterisk-us...@ics-il.net wrote: My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com I think you need something inside [DID-incoming] like for example... exten = s,1,NoOP(-incoming call---) exten = s,n,Playback(wellcome) # Looking for s in DID-incoming (domain 208.100.1.33) # Reliably Transmitting (no NAT) to 208.1.87.235:5060: # SIP/2.0 404 Not Found -- -- Adrià Vidal adriavi...@gmail.com ___ -- 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] ATA gateway with 2 ethernet interfaces
Agg, I felt bad about being pedantic. How about splitting the load and reducing the single point of failure? Instead of one big ATA how about a number of smaller ones (two port) split between your switches? j On Fri, 16 Jan 2009, Jeff LaCoursiere wrote: I would be more worried about the ATA gateway failing than the switch, as you have found yourself. How about two gateways and two phones on everyone's desk :) j On Fri, 16 Jan 2009, Adam Moffett wrote: I don't know of any ATA like that except the grandstream. The service provider grade way to do this would probably be a Cisco (or similar) with a T1 interface and a channel bank to break the T1 into 24 FXS ports. Hi, I'm looking for an 8+ FXS ATA gateway (at least 8 ports but preferably at most 24 ports) with 2 ethernet interfaces for network/switch redundancy. So far I've only found the Grandstream GXW4008. I've searched similar brands such as Linksys and higher-end brands such as Quintum, but they all seem to have just one NIC. So, if the switch the ATA is connected to fails then I'm out of business (at least until I replace the switch but that's usually too long for a busy system). The GXW4008 device is very useful for this scenario. It has 2 RJ45 ports (called WAN and LAN) and I've set them up in two local subnets. Not only does the ATA keep working without human intervention if one of the switches goes down but if both switches are up it can load balance between the two (simply by using DNS SRV with the same weights). Unfortunately, Grandstream in general doesn't seem to be very reliable although the latest GXW4008 firmware has proven to be quite stable in my case (previous releases were buggy). So I'm looking for alternatives to the GXW4008, even if it has to cost me more money. Does anyone know of an 8+ FXS ATA brand/model with 2 ethernet interfaces? Thanks in advance. Vieri ___ -- 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 ___ -- 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 ___ -- 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 ___ -- 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] pstn hangs up: MWI no message waiting ??
Doug Bailey wrote: - sean darcy seandar...@gmail.com wrote: pstn incoming on a TDM400P, sometimes i* won't answer, going into a loop like this: -- Starting simple switch on 'DAHDI/4-1' [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 18 (Ring Begin)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 (Ring/Answered)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: Channel 4 no message waiting! This message occurs when the pstn sends a FSK spill indicating the message waiting status of the FXO port in question. This may encoded in the caller ID indicator or may be contained in its own message spill. This is output as a NOTICE logging message. Regards, Doug Bailey I'm not sure I understand all that, but why does asterisk hang up? It means I can't receive any calls on that pstn line. AFAICS, only restarting asterisk allows calls to be received. A cron job to restart every 5 minutes? sean ___ -- 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] pstn hangs up: MWI no message waiting ??
On Fri, Jan 16, 2009 at 4:10 PM, sean darcy seandar...@gmail.com wrote: Doug Bailey wrote: - sean darcy seandar...@gmail.com wrote: pstn incoming on a TDM400P, sometimes i* won't answer, going into a loop like this: -- Starting simple switch on 'DAHDI/4-1' [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 18 (Ring Begin)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 (Ring/Answered)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: Channel 4 no message waiting! This message occurs when the pstn sends a FSK spill indicating the message waiting status of the FXO port in question. This may encoded in the caller ID indicator or may be contained in its own message spill. This is output as a NOTICE logging message. Regards, Doug Bailey I'm not sure I understand all that, but why does asterisk hang up? It means I can't receive any calls on that pstn line. AFAICS, only restarting asterisk allows calls to be received. A cron job to restart every 5 minutes? sean ___ -- 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 I hear Zaptel is pretty stable.. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] pstn hangs up: MWI no message waiting ??
Yes, BUT .. not 100% and discontinued in 1.4.22 on ... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Friday, January 16, 2009 3:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] pstn hangs up: MWI no message waiting ?? On Fri, Jan 16, 2009 at 4:10 PM, sean darcy seandar...@gmail.com wrote: Doug Bailey wrote: - sean darcy seandar...@gmail.com wrote: pstn incoming on a TDM400P, sometimes i* won't answer, going into a loop like this: -- Starting simple switch on 'DAHDI/4-1' [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 18 (Ring Begin)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 (Ring/Answered)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: Channel 4 no message waiting! This message occurs when the pstn sends a FSK spill indicating the message waiting status of the FXO port in question. This may encoded in the caller ID indicator or may be contained in its own message spill. This is output as a NOTICE logging message. Regards, Doug Bailey I'm not sure I understand all that, but why does asterisk hang up? It means I can't receive any calls on that pstn line. AFAICS, only restarting asterisk allows calls to be received. A cron job to restart every 5 minutes? sean ___ -- 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 I hear Zaptel is pretty stable.. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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 ___ -- 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] pstn hangs up: MWI no message waiting ??
Why not do a zap restart instead of restarting asterisk? You could write an AGI to do the ZR when the condition occurred and lines where empty. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Friday, January 16, 2009 3:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] pstn hangs up: MWI no message waiting ?? On Fri, Jan 16, 2009 at 4:10 PM, sean darcy seandar...@gmail.com wrote: Doug Bailey wrote: - sean darcy seandar...@gmail.com wrote: pstn incoming on a TDM400P, sometimes i* won't answer, going into a loop like this: -- Starting simple switch on 'DAHDI/4-1' [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 18 (Ring Begin)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 (Ring/Answered)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: Channel 4 no message waiting! This message occurs when the pstn sends a FSK spill indicating the message waiting status of the FXO port in question. This may encoded in the caller ID indicator or may be contained in its own message spill. This is output as a NOTICE logging message. Regards, Doug Bailey I'm not sure I understand all that, but why does asterisk hang up? It means I can't receive any calls on that pstn line. AFAICS, only restarting asterisk allows calls to be received. A cron job to restart every 5 minutes? sean ___ -- 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 I hear Zaptel is pretty stable.. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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 ___ -- 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] pstn hangs up: MWI no message waiting ??
Sounds like an awful hack. What does DAHDI do that Zaptel does not? Sounds more like a post for the bugs list On Fri, Jan 16, 2009 at 4:48 PM, Danny Nicholas da...@debsinc.com wrote: Why not do a zap restart instead of restarting asterisk? You could write an AGI to do the ZR when the condition occurred and lines where empty. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Friday, January 16, 2009 3:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] pstn hangs up: MWI no message waiting ?? On Fri, Jan 16, 2009 at 4:10 PM, sean darcy seandar...@gmail.com wrote: Doug Bailey wrote: - sean darcy seandar...@gmail.com wrote: pstn incoming on a TDM400P, sometimes i* won't answer, going into a loop like this: -- Starting simple switch on 'DAHDI/4-1' [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 18 (Ring Begin)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 (Ring/Answered)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: Channel 4 no message waiting! This message occurs when the pstn sends a FSK spill indicating the message waiting status of the FXO port in question. This may encoded in the caller ID indicator or may be contained in its own message spill. This is output as a NOTICE logging message. Regards, Doug Bailey I'm not sure I understand all that, but why does asterisk hang up? It means I can't receive any calls on that pstn line. AFAICS, only restarting asterisk allows calls to be received. A cron job to restart every 5 minutes? sean I hear Zaptel is pretty stable.. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] pstn hangs up: MWI no message waiting ??
So Zap restart is a worse hack than restarting *? What DAHDI does that Zaptel doesn't - 1. Makes Digium lawyers happy 2. Gives Developers headaches 3. Makes Coffee and soft drink manufacturers happy since we need so much caffine. 4. Make priests happy since we have to go to confession about it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Friday, January 16, 2009 3:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] pstn hangs up: MWI no message waiting ?? Sounds like an awful hack. What does DAHDI do that Zaptel does not? Sounds more like a post for the bugs list On Fri, Jan 16, 2009 at 4:48 PM, Danny Nicholas da...@debsinc.com wrote: Why not do a zap restart instead of restarting asterisk? You could write an AGI to do the ZR when the condition occurred and lines where empty. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Friday, January 16, 2009 3:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] pstn hangs up: MWI no message waiting ?? On Fri, Jan 16, 2009 at 4:10 PM, sean darcy seandar...@gmail.com wrote: Doug Bailey wrote: - sean darcy seandar...@gmail.com wrote: pstn incoming on a TDM400P, sometimes i* won't answer, going into a loop like this: -- Starting simple switch on 'DAHDI/4-1' [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 18 (Ring Begin)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 (Ring/Answered)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: Channel 4 no message waiting! This message occurs when the pstn sends a FSK spill indicating the message waiting status of the FXO port in question. This may encoded in the caller ID indicator or may be contained in its own message spill. This is output as a NOTICE logging message. Regards, Doug Bailey I'm not sure I understand all that, but why does asterisk hang up? It means I can't receive any calls on that pstn line. AFAICS, only restarting asterisk allows calls to be received. A cron job to restart every 5 minutes? sean I hear Zaptel is pretty stable.. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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 ___ -- 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] pstn hangs up: MWI no message waiting ??
Both are hacks. Neither are good hacks. On Fri, Jan 16, 2009 at 5:04 PM, Danny Nicholas da...@debsinc.com wrote: So Zap restart is a worse hack than restarting *? What DAHDI does that Zaptel doesn't - 1. Makes Digium lawyers happy 2. Gives Developers headaches 3. Makes Coffee and soft drink manufacturers happy since we need so much caffine. 4. Make priests happy since we have to go to confession about it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Friday, January 16, 2009 3:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] pstn hangs up: MWI no message waiting ?? Sounds like an awful hack. What does DAHDI do that Zaptel does not? Sounds more like a post for the bugs list On Fri, Jan 16, 2009 at 4:48 PM, Danny Nicholas da...@debsinc.com wrote: Why not do a zap restart instead of restarting asterisk? You could write an AGI to do the ZR when the condition occurred and lines where empty. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Friday, January 16, 2009 3:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] pstn hangs up: MWI no message waiting ?? On Fri, Jan 16, 2009 at 4:10 PM, sean darcy seandar...@gmail.com wrote: Doug Bailey wrote: - sean darcy seandar...@gmail.com wrote: pstn incoming on a TDM400P, sometimes i* won't answer, going into a loop like this: -- Starting simple switch on 'DAHDI/4-1' [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 18 (Ring Begin)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 (Ring/Answered)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: Channel 4 no message waiting! This message occurs when the pstn sends a FSK spill indicating the message waiting status of the FXO port in question. This may encoded in the caller ID indicator or may be contained in its own message spill. This is output as a NOTICE logging message. Regards, Doug Bailey I'm not sure I understand all that, but why does asterisk hang up? It means I can't receive any calls on that pstn line. AFAICS, only restarting asterisk allows calls to be received. A cron job to restart every 5 minutes? sean I hear Zaptel is pretty stable.. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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 ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Snom 300 vs Grandstream gxp
On Fri, 2009-01-16 at 15:52 +, Julian Lyndon-Smith wrote: Can anyone who has used both comment on the pros and cons ? Need to buy about 30 of these, for a small company with limited IT support. For evaluation a got a couple of different phones. gxp2000, cheap, but it works, some people (not me) have lots of problems with them. 4-sip-users per phone. no VLAN support. GXP-budget, even cheaper, but works, just one single sip-user per phone. snom320, bit more expensive, also works great, easy to upgrade. Latest version even got v6-support. all versions have VLAN support, 6-sip-users per phone. Easy to configure with web-interface. Siemens optipoint 410, something to avoid! (even it it were less expensive) If power/lan are plugged in wrong order, you get an instant brick, as the firmware is wiped and no possibility to get it back. I mention VLAN support, in case you don't have separate lan's it might be worthwhile if a lot of data passes through the rest of the lan hw ___ -- 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] UpdateConfig : Appending line fails
Hello list, Can someone please point me out why would a stream like the following only write ONE line (the first) on the given file? Action: login Username: test Secret: 123456 Action: UpdateConfig SrcFilename: voicemail2.conf DstFilename: voicemail2.conf Action-00: Append Cat-00: default Var-00: 127 Value-00: , Jason Bourne97, jaso...@nocia.gov.do ActionID: 1256187957 Action: UpdateConfig SrcFilename: voicemail2.conf DstFilename: voicemail2.conf Action-01: Append Cat-01: default Var-01: 125 Value-01: 5, Jason Bourne76, jaso...@nocia.gov.do ActionID: 1607673137 Action: UpdateConfig SrcFilename: voicemail2.conf DstFilename: voicemail2.conf Action-02: Append Cat-02: default Var-02: 122 Value-02: , Jason Bourne74, jaso...@nocia.gov.do ActionID: 165797792 Action: UpdateConfig SrcFilename: voicemail2.conf DstFilename: voicemail2.conf Action-03: Append Cat-03: default Var-03: 128 Value-03: , Jason Bourne48, jaso...@nocia.gov.do ActionID: 1743636529 Action: UpdateConfig SrcFilename: voicemail2.conf DstFilename: voicemail2.conf Action-04: Append Cat-04: default Var-04: 126 Value-04: 5, Jason Bourne18, jaso...@nocia.gov.do ActionID: 495446608 Action: Logoff Username: test I'm I missing something? PS. (I receive 'success' for all of the updating attempts) Thanks in advice, -- Jose P. Espinal http://blog.slackware-es.com ___ -- 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] CDR problems -- two call legs create only one CDR. Using ForkCDR() not even working.
On Fri, Jan 16, 2009 at 6:12 PM, Örn Arnarson o...@arnarson.net wrote: Am I missing something? Any ideas appreciated. No you are not missing anything. The Asterisk CDR implementation has a number of issues and one CDR per bridge is one of them. There is currently a re-design discussion going on on the list at the moment. If you're really interested you can read the new design document at http://svn.digium.com/svn/asterisk/team/murf/RFCs/CDRfix2.rfc.txt. Regards, Greyman. ___ -- 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] pstn hangs up: MWI no message waiting ??
Danny Nicholas wrote: Why not do a zap restart instead of restarting asterisk? You could write an AGI to do the ZR when the condition occurred and lines where empty. Yes, a cron job to restart zaptel would cut off any call then existing. But how would I test for it? I can imagine: exten=s,n,ExecIf(some damn thing, System(service dahdi restart)) It's the some damn thing I can't imagine. How do you test if dahdi is acting up? sean ___ -- 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] UpdateConfig : Appending line fails
On Friday 16 January 2009 16:47:40 Jose P. Espinal wrote: Can someone please point me out why would a stream like the following only write ONE line (the first) on the given file? Action: login Username: test Secret: 123456 Action: UpdateConfig SrcFilename: voicemail2.conf DstFilename: voicemail2.conf Action-00: Append Cat-00: default Var-00: 127 Value-00: , Jason Bourne97, jaso...@nocia.gov.do ActionID: 1256187957 This is a single valid command, and so it works. Action: UpdateConfig SrcFilename: voicemail2.conf DstFilename: voicemail2.conf Action-01: Append Cat-01: default Var-01: 125 Value-01: 5, Jason Bourne76, jaso...@nocia.gov.do ActionID: 1607673137 This command did not specify an Action-00. Action: UpdateConfig SrcFilename: voicemail2.conf DstFilename: voicemail2.conf Action-02: Append Cat-02: default Var-02: 122 Value-02: , Jason Bourne74, jaso...@nocia.gov.do ActionID: 165797792 This command did not specify an Action-00. Action: UpdateConfig SrcFilename: voicemail2.conf DstFilename: voicemail2.conf Action-03: Append Cat-03: default Var-03: 128 Value-03: , Jason Bourne48, jaso...@nocia.gov.do ActionID: 1743636529 This command did not specify an Action-00. Action: UpdateConfig SrcFilename: voicemail2.conf DstFilename: voicemail2.conf Action-04: Append Cat-04: default Var-04: 126 Value-04: 5, Jason Bourne18, jaso...@nocia.gov.do ActionID: 495446608 This command did not specify an Action-00. Action: Logoff Username: test I'm I missing something? Perhaps you actually meant to do the following command: Action: UpdateConfig SrcFilename: voicemail2.conf DstFilename: voicemail2.conf Action-00: Append Cat-00: default Var-00: 127 Value-00: , Jason Bourne97, jaso...@nocia.gov.do Action-01: Append Cat-01: default Var-01: 125 Value-01: 5, Jason Bourne76, jaso...@nocia.gov.do Action-02: Append Cat-02: default Var-02: 122 Value-02: , Jason Bourne74, jaso...@nocia.gov.do Action-03: Append Cat-03: default Var-03: 128 Value-03: , Jason Bourne48, jaso...@nocia.gov.do Action-04: Append Cat-04: default Var-04: 126 Value-04: 5, Jason Bourne18, jaso...@nocia.gov.do ActionID: 495446608 -- Tilghman ___ -- 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] pstn hangs up: MWI no message waiting ??
On Friday 16 January 2009 17:43:21 sean darcy wrote: Danny Nicholas wrote: Why not do a zap restart instead of restarting asterisk? You could write an AGI to do the ZR when the condition occurred and lines where empty. Yes, a cron job to restart zaptel would cut off any call then existing. But how would I test for it? I can imagine: exten=s,n,ExecIf(some damn thing, System(service dahdi restart)) It's the some damn thing I can't imagine. How do you test if dahdi is acting up? Not a service restart, but a dahdi restart. You can't restart the dahdi service without first stopping Asterisk, anyway. if [ `../asterisk-trunk/contrib/scripts/astcli core show channels | wc -l` = 3 ]; then asterisk -rx 'dahdi restart'; fi -- Tilghman ___ -- 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] pstn hangs up: MWI no message waiting ??
Tilghman Lesher wrote: On Friday 16 January 2009 17:43:21 sean darcy wrote: Danny Nicholas wrote: Why not do a zap restart instead of restarting asterisk? You could write an AGI to do the ZR when the condition occurred and lines where empty. Yes, a cron job to restart zaptel would cut off any call then existing. But how would I test for it? I can imagine: exten=s,n,ExecIf(some damn thing, System(service dahdi restart)) It's the some damn thing I can't imagine. How do you test if dahdi is acting up? Not a service restart, but a dahdi restart. You can't restart the dahdi service without first stopping Asterisk, anyway. if [ `../asterisk-trunk/contrib/scripts/astcli core show channels | wc -l` = 3 ]; then asterisk -rx 'dahdi restart'; fi Wow. I'll try that tomorrow. Put it as the cmd right after answer(), right? Or maybe, h,1 ? Well anyway, at least I'll be able to receive calls over pstn with dahdi. Thanks. sean ___ -- 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] CDR Rewrite -- Questions to the users
On Sat, Jan 17, 2009 at 3:08 AM, Steve Murphy m...@digium.com wrote: Greyman-- I've been thinking of Benny Amorsen's comments on Simple CDRs... This is tricky... I need to create these CDR's for the billing system: src: A start: e1 ans: e2 end: e6 dst: B disp: ANSW src: A start: e4 ans: e5 end: e6 dst: B disp: ANSW If I do the same substitution again, I get this: A-B start: e1 ans: e2 end: e4. Whoops, end time is wrong. A-C start: e1 ans: e5 end: e4. Whoops, both start and end times are wrong. CDR2 needs to find e1 so it can replace start, while CDR3 shouldn't have anything replaced. I can't think of a query which will do this correctly. Do you have any ideas that might make this work for him? I guess simple CDR's won't work for everyone, but... if they could work in this case From what I can gather the problem Benny has is getting the src field correct for a transfer. In the traditional Telco World the src (or A Number) field tends to be both the callerid of the customer and an identifier that ties the CDR to the customer for billing purposes. With Asterisk and a lot of other modern day softswitches there's usually a field called accountcode or similar which can be used to tie a CDR to a customer. The src field is then only fulfilling one role which is to hold the callerid that was set or recieved for the call. The trick with transfers is to forget about the src field for billing purposes and make sure the accountcode for the call is set in accordance with the business rules. For example if two customers A and B are talking to each other and A blind transfers B to a billable destination Z then who pays for the call from B to Z? There is no right answer but as far as the CDRs are concerned it's irrelvant as long as each call is recorded and the accountcode can be set within the dialplan both choices can be accomodated. With the simple CDR approach it could end up that there are multiple CDRs for a customer for the one call since they could be charged for both ends of a transfer. I don't see that as an issue. Regards, Greyman. ___ -- 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] UpdateConfig : Appending line fails
Thank you very much, Tilghman, I did not respond before because for some reason the IP of the list server was blacklisted on spamhaus.org and I was not getting the messages see: 2009-01-16 22:47:49 H=(lists.digium.com) [216.207.245.1] F=asterisk-users-boun...@lists.digium.com rejected RCPT r...@dangerclan.net: JunkMail rejected - (lists.digium.com) [216.207.245.1] is in an RBL, see http://www.spamhaus.org/query/bl?ip=216.207.245.1; About UpdateConfig syntax, how did you find out the correct way of sending various sets of parameters? I was looking in google, the ATFOT v2 Book, and nothing showed up. Thanks again, regards. -- Jose P. Espinal / // // I'm I missing something? / Perhaps you actually meant to do the following command: Action: UpdateConfig SrcFilename: voicemail2.conf DstFilename: voicemail2.conf Action-00: Append Cat-00: default Var-00: 127 Value-00: , Jason Bourne97, jason25 at noCia.gov.do http://lists.digium.com/mailman/listinfo/asterisk-users Action-01: Append Cat-01: default Var-01: 125 Value-01: 5, Jason Bourne76, jason12 at noCia.gov.do http://lists.digium.com/mailman/listinfo/asterisk-users Action-02: Append Cat-02: default Var-02: 122 Value-02: , Jason Bourne74, jason49 at noCia.gov.do http://lists.digium.com/mailman/listinfo/asterisk-users Action-03: Append Cat-03: default Var-03: 128 Value-03: , Jason Bourne48, jason45 at noCia.gov.do http://lists.digium.com/mailman/listinfo/asterisk-users Action-04: Append Cat-04: default Var-04: 126 Value-04: 5, Jason Bourne18, jason64 at noCia.gov.do http://lists.digium.com/mailman/listinfo/asterisk-users ActionID: 495446608 -- Tilghman ___ -- 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] Portech MV-378 with Asterisk
Marco, The configs work fine for me. I can receive calls with no problem. Now, were you able to dial using the sim card? I cant figure out how I can do it since asterisk doesnt have a channel to place call through the portech gateway. On Fri, Jan 16, 2009 at 12:04 PM, Pascal Bruno tipas...@gmail.com wrote: Thank you!, I will try that in a few hours and let you know what happens. On Fri, Jan 16, 2009 at 11:01 AM, Marco Signorini marcota...@libero.itwrote: Pascal Bruno wrote: Thanks for your reply! Can you tell me what you have in your Portech configuration settings (Mobile to Lan Settings; Sip Proxy settings etc...) My sip.conf file is pretty similar to yours but still cant register. On Fri, Jan 16, 2009 at 3:47 AM, Marco Signorini marcota...@libero.itwrote: Emmanuel Pascal Bruno wrote: Has anyone been able to configure portech's mv-378 gateway with asterisk? I did the configuration as per the manual but it does not work. My server sees the portech gateway, but when the gateway is trying to register to my server it fails. It says peer is not suppose to register. The gateway and the asterisk box are on two different location (two network, 2 differrent IP address). I would appreciate any kind of tutorial or advice on how to make it work. Thanks -- ___ -- 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 Hi, I've an installation working with Portech MV-370. I'm supposing it's quite similar to what you have. If it could be useful to you, this is my sip.conf configuration file. [GSMGtw1] type=friend context=from-gsm host=dynamic; we have a DHCP assigned address secret=reallyverysecret nat=no ; there is not NAT between phone and Asterisk canreinvite=no dtmfmode=INFO insecure=invite ; required to overcome authentication problems in incoming calls call-limit=1 ; permit only 1 outgoing call at a time disallow=all allow=ulaw allow=alaw allow=gsm qualify=500 I remember that I've found a bug on the firmware that prevents to the unit to register correctly on my asterisk box unless I'm using the raw IP address instead of the name of the asterisk box. I remember something wrong in cryptography chiper/dechiper based on realm... So, if you have problems, let's try to specify the asterisk raw IP address in the Portech. Best regards, Marco Signorini. Hi, I don't know if the problem could be in the Mobile to Lan or Lan to Mobile settings because these settings are related on how calls coming from/to mobile are routed. I didn't use the Portech routing features at all because I need a simple GSM gateway to/from the asterisk box. For this reason: 1. The only rule I've on Mobile to Lan is CID=*; url=...@192.168.0.5where mob is the extension I've generated in the asterisk box under the context where the Portech operates; 2. The only rule I've on Lan to Mobile is URL=*; Call Num=# I think the most relevant parameters for your problem are under the Service Domain menu option (assuming that the firmware you have is similar to what I've). On this menu I've compiled the 1st Realm (as I've only one account) like that: UserName: GSMGtw1 RegisterName: GSMGtw1 RegisterPassword: reallyverysecret Domain Server: 192.168.0.5 Proxy Server: 192.168.0.5 Pay attention that, having specified the Domain Server with the raw IP address, asterisk needs to be able to authenticate peers associated to that. For this reason I've set: domain=192.168.0.5 on sip.conf [general] section (remember to issue a sip reload from asterisk cli). Hope this helps! Best regards. Marco Signorini Marco Signorini INGEGNI Tech S.r.l. http://www.ingegnitech.com ___ -- 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 ___ -- 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] Portech MV-378 with Asterisk
I want to dial out using the sim card. What I did, I have used the SIP channel ex: Channel: SIP/thenum...@mv378 It shows the called is being made in the dialplan, but the number I have entered does not dial, it just goes straight to the specified dialplan extensions. Then what I did, in the Lan to Mobile Table, I put * in url and the number I wanted to dial in call num, then the call was made to that number using the sim card properly. I was wondering if I cannot supply the number to be dialed using an asterisk call file, or do I have to put that number in the Lan to Mobile table. Any help would be appreciated. Thanks On Sat, Jan 17, 2009 at 12:39 AM, Pascal Bruno tipas...@gmail.com wrote: Marco, The configs work fine for me. I can receive calls with no problem. Now, were you able to dial using the sim card? I cant figure out how I can do it since asterisk doesnt have a channel to place call through the portech gateway. On Fri, Jan 16, 2009 at 12:04 PM, Pascal Bruno tipas...@gmail.com wrote: Thank you!, I will try that in a few hours and let you know what happens. On Fri, Jan 16, 2009 at 11:01 AM, Marco Signorini marcota...@libero.itwrote: Pascal Bruno wrote: Thanks for your reply! Can you tell me what you have in your Portech configuration settings (Mobile to Lan Settings; Sip Proxy settings etc...) My sip.conf file is pretty similar to yours but still cant register. On Fri, Jan 16, 2009 at 3:47 AM, Marco Signorini marcota...@libero.itwrote: Emmanuel Pascal Bruno wrote: Has anyone been able to configure portech's mv-378 gateway with asterisk? I did the configuration as per the manual but it does not work. My server sees the portech gateway, but when the gateway is trying to register to my server it fails. It says peer is not suppose to register. The gateway and the asterisk box are on two different location (two network, 2 differrent IP address). I would appreciate any kind of tutorial or advice on how to make it work. Thanks -- ___ -- 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 Hi, I've an installation working with Portech MV-370. I'm supposing it's quite similar to what you have. If it could be useful to you, this is my sip.conf configuration file. [GSMGtw1] type=friend context=from-gsm host=dynamic; we have a DHCP assigned address secret=reallyverysecret nat=no ; there is not NAT between phone and Asterisk canreinvite=no dtmfmode=INFO insecure=invite ; required to overcome authentication problems in incoming calls call-limit=1 ; permit only 1 outgoing call at a time disallow=all allow=ulaw allow=alaw allow=gsm qualify=500 I remember that I've found a bug on the firmware that prevents to the unit to register correctly on my asterisk box unless I'm using the raw IP address instead of the name of the asterisk box. I remember something wrong in cryptography chiper/dechiper based on realm... So, if you have problems, let's try to specify the asterisk raw IP address in the Portech. Best regards, Marco Signorini. Hi, I don't know if the problem could be in the Mobile to Lan or Lan to Mobile settings because these settings are related on how calls coming from/to mobile are routed. I didn't use the Portech routing features at all because I need a simple GSM gateway to/from the asterisk box. For this reason: 1. The only rule I've on Mobile to Lan is CID=*; url=...@192.168.0.5where mob is the extension I've generated in the asterisk box under the context where the Portech operates; 2. The only rule I've on Lan to Mobile is URL=*; Call Num=# I think the most relevant parameters for your problem are under the Service Domain menu option (assuming that the firmware you have is similar to what I've). On this menu I've compiled the 1st Realm (as I've only one account) like that: UserName: GSMGtw1 RegisterName: GSMGtw1 RegisterPassword: reallyverysecret Domain Server: 192.168.0.5 Proxy Server: 192.168.0.5 Pay attention that, having specified the Domain Server with the raw IP address, asterisk needs to be able to authenticate peers associated to that. For this reason I've set: domain=192.168.0.5 on sip.conf [general] section (remember to issue a sip reload from asterisk cli). Hope this helps! Best regards. Marco Signorini Marco Signorini INGEGNI Tech S.r.l. http://www.ingegnitech.com ___ -- 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] Portech MV-378 with Asterisk
Sorry for bothering you, but I got it, I just had to put # in callnum! On Sat, Jan 17, 2009 at 1:44 AM, Pascal Bruno tipas...@gmail.com wrote: I want to dial out using the sim card. What I did, I have used the SIP channel ex: Channel: SIP/thenum...@mv378 It shows the called is being made in the dialplan, but the number I have entered does not dial, it just goes straight to the specified dialplan extensions. Then what I did, in the Lan to Mobile Table, I put * in url and the number I wanted to dial in call num, then the call was made to that number using the sim card properly. I was wondering if I cannot supply the number to be dialed using an asterisk call file, or do I have to put that number in the Lan to Mobile table. Any help would be appreciated. Thanks On Sat, Jan 17, 2009 at 12:39 AM, Pascal Bruno tipas...@gmail.com wrote: Marco, The configs work fine for me. I can receive calls with no problem. Now, were you able to dial using the sim card? I cant figure out how I can do it since asterisk doesnt have a channel to place call through the portech gateway. On Fri, Jan 16, 2009 at 12:04 PM, Pascal Bruno tipas...@gmail.comwrote: Thank you!, I will try that in a few hours and let you know what happens. On Fri, Jan 16, 2009 at 11:01 AM, Marco Signorini marcota...@libero.itwrote: Pascal Bruno wrote: Thanks for your reply! Can you tell me what you have in your Portech configuration settings (Mobile to Lan Settings; Sip Proxy settings etc...) My sip.conf file is pretty similar to yours but still cant register. On Fri, Jan 16, 2009 at 3:47 AM, Marco Signorini marcota...@libero.itwrote: Emmanuel Pascal Bruno wrote: Has anyone been able to configure portech's mv-378 gateway with asterisk? I did the configuration as per the manual but it does not work. My server sees the portech gateway, but when the gateway is trying to register to my server it fails. It says peer is not suppose to register. The gateway and the asterisk box are on two different location (two network, 2 differrent IP address). I would appreciate any kind of tutorial or advice on how to make it work. Thanks -- ___ -- 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 Hi, I've an installation working with Portech MV-370. I'm supposing it's quite similar to what you have. If it could be useful to you, this is my sip.conf configuration file. [GSMGtw1] type=friend context=from-gsm host=dynamic; we have a DHCP assigned address secret=reallyverysecret nat=no ; there is not NAT between phone and Asterisk canreinvite=no dtmfmode=INFO insecure=invite ; required to overcome authentication problems in incoming calls call-limit=1 ; permit only 1 outgoing call at a time disallow=all allow=ulaw allow=alaw allow=gsm qualify=500 I remember that I've found a bug on the firmware that prevents to the unit to register correctly on my asterisk box unless I'm using the raw IP address instead of the name of the asterisk box. I remember something wrong in cryptography chiper/dechiper based on realm... So, if you have problems, let's try to specify the asterisk raw IP address in the Portech. Best regards, Marco Signorini. Hi, I don't know if the problem could be in the Mobile to Lan or Lan to Mobile settings because these settings are related on how calls coming from/to mobile are routed. I didn't use the Portech routing features at all because I need a simple GSM gateway to/from the asterisk box. For this reason: 1. The only rule I've on Mobile to Lan is CID=*; url=...@192.168.0.5where mob is the extension I've generated in the asterisk box under the context where the Portech operates; 2. The only rule I've on Lan to Mobile is URL=*; Call Num=# I think the most relevant parameters for your problem are under the Service Domain menu option (assuming that the firmware you have is similar to what I've). On this menu I've compiled the 1st Realm (as I've only one account) like that: UserName: GSMGtw1 RegisterName: GSMGtw1 RegisterPassword: reallyverysecret Domain Server: 192.168.0.5 Proxy Server: 192.168.0.5 Pay attention that, having specified the Domain Server with the raw IP address, asterisk needs to be able to authenticate peers associated to that. For this reason I've set: domain=192.168.0.5 on sip.conf [general] section (remember to issue a sip reload from asterisk cli). Hope this helps! Best regards. Marco Signorini Marco Signorini INGEGNI Tech S.r.l. http://www.ingegnitech.com ___ -- Bandwidth and Colocation
Re: [asterisk-users] mini-PCI FXS card?
The EdgePBX FX08 has up to eight ports (using Digium/compatible modules), a couple of ethernet interfaces, and runs Astfin2 (Asterisk 1.4.21 and uClinux 2.6.22 on a Blackfin DSP). http://www.edgepbx.cn/shop/index.php?controller=productproduct_id=6 or you might prefer its baby brother, the FX02, with up to two ports, and one ethernet interface: http://www.edgepbx.cn/shop/index.php?controller=productpath=19product_id=1 I recently purchased a FX08 for home use, and have been having a blast with it. I bought it with one FxO and one FxS module, then added some Digium modules I had from a TDM400. -- Paul Adam Moffett wrote: Is there any product that's a single port mini-PCI FXS card? I'm aware of the Openvox A400M http://www.openvox.com.cn/products.php?genre_id=39, but I really only wanted one port. How about a single or dual port PCI or PCI express FXS card? Basically I wanted to build a small linux router with one or two phone ports. Alternatively, is there already a router or single board computer with FXS ports that I could run linux/asterisk on? ___ -- 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 ___ -- 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] Asterisk Upgrade
On Fri, Jan 16, 2009 at 01:24:16PM +, Gordon Henderson wrote: On Fri, 16 Jan 2009, Alex Balashov wrote: 1. rm -rf /var/lib/asterisk /var/spool/asterisk /etc/asterisk I'd suggest not removing /etc/asterisk if that's the only source of your config files... If you (re)generate them from elsewhere, it's probably OK. /var/lib/asterisk/sounds may have some custom sound files . /var/spool/asterisk/voicemail has voicemail and voicemail prompts. and the important one, I'd have thought is /usr/lib/asterisk/modules -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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
