Re: [Asterisk-Users] Asterisk and Ericsson PBX
I am unsure of what you want to achieve. Do you want to interconnect BP and Asterisk, or replace BP with Asterisk? What is the purpose of proprietary software you mention? Please give more details. Niksa [EMAIL PROTECTED] wrote: Hi, I´m trying to migrate my propietary software to an asterisk server connected to a Ericsson BP 128i PBX. I´ve been looking at the asterisk web, user forums, published docs about how to use the PBX as the hardware device but I haven´t found anything. I think this is possible. The old server is currently connected to the Ericcson via serial port. Please, help. Thanks a lot. This message is for the designated recipient only and may contain privileged, proprietary, or otherwise private information. If you have received it in error, please notify the sender immediately and delete the original. Any other use of the email by you is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound card Line-In as MOH source
Hi, this is a rather ugly solution I devised. Create a script called 'ast-playlinein' (or whatever) in /usr/sbin, as follows: #!/bin/bash /usr/bin/arecord -q -c 1 -r 8000 --buffer-size=2048 -f S16_LE -D hw:0,0 -t raw In musiconhold.conf: [classes] default = custom:/var/lib/asterisk/mohmp3,/usr/sbin/ast-playlinein And that's it. Basically, the script takes input from line-in and sends it to standard output in the format Asterisk expects. You also have to select line-in as recording source using alsamixer utility. If you are using a sound system other than ALSA, odds are that it has some tool similar to arecord. Bear in mind that /var/lib/asterisk/mohmp3 directory still has to contain at least one file with .mp3 extension (can be whatever you like). Hope this helps. -- Niksa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Avaya 4602 SIP phone
I tested Avaya 4602 a while ago and I successfully instaled SIP software and connected the phone to Asterisk. I remember that the upgrade procedure was a bit complicated (includes TFTP and Web server as I recall), but if you follow it to the letter you should have no problems. If the phone keeps rebooting, my guess is that it cannot connect to the Web server, or it cannot access the files on the server. However, this phone is designed specifically for proprietary Avaya platform, and I couldn't achieve anything but the most basic telephony functions. Therefore, I concluded that this phone is not suitable for Asterisk, so my suggestion is to try some other phone. Niksa Yao, Yuanbin wrote: Hi, I have been trying to connect Avaya 4602 SIP phone to Asterisk, but the phone keeps rebooting after I downloaded the SIP software (Avaya phone release 050205). I would like to know if anyone succeeded to hook up Avaya SIP phone to Asterisk. I appreciate your help! Regards, yyao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + GNUGK
Assuming your h.323 phones are registered with gnugk, you need to instruct gnugk to forward certain numbers to Asterisk. In OH323 (which I am using) you would need to add something like: [register] gwprefix=0 gwprefix=1 etc. In h323.conf, I believe you have to add prefix=xxx in your endpoint definition. Bear in mind though that H.323 support in Asterisk is rather inadequate (only basic telephony functions are available). Niksa Baldun Ganbold Tsagaankhuu wrote: Hi, I'm trying to configure asterisk to work with gnugk-2.0.8. Something like: SIP phones - ASTERISK - GNUGK -Cisco GW - PSTN | h323 phones Following is h323.conf: [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=g729 gatekeeper = x.x.x.x secret = 1234 AllowGKRouted = yes noFastStart = yes noH245Tunneling = yes noSilenceSuppression = yes [30598272] type=h323 prefix=111,115,116,117 context=home ;e164=117 [115] type=user context=home incominglimit=4 sip.conf [general] port=5060 ; Port to bind to bindaddr=0.0.0.0; Address to bind SIP channel to context=home; Default context for incoming calls musicclass=default ;videosupport=yes allow=g729 allow=g723 ;externip = 202.179.0.164 ;localnet=192.168.0.0/255.255.0.0 [111] type=friend username=111 ;secret= host=dynamic nat=yes defaultip=192.168.0.11 context=home canreinvite=no callerid=111 [EMAIL PROTECTED] [112] type=friend username=112 ;secret= nat=yes host=dynamic context=home canreinvite=no callerid=112 [EMAIL PROTECTED] [115] type=friend username=115 ;secret=1234 defaultip=192.168.0.62 nat=yes host=dynamic context=home canreinvite=no callerid=115 [EMAIL PROTECTED] [116] type=friend username=116 ;secret=4321 host=dynamic context=home canreinvite=no callerid=116 [EMAIL PROTECTED] As in above configuration I'm registering Asterisk as an endpoint to gnugk. It is working and I can make calls from SIP phones to PSTN. However my question is, how can I call from h323 endpoints to SIP phones or vice versa in above case? Is it possible? I'm afraid, it can't since asterisk is itself an one endpoint to gnugk. If possible how can I make it work? If not, is it possible to register or make each SIP phones to be known to gnugk? How can I accomplish that? Ideally this solution could be the best. It would be very helpful if somebody can show me the config samples. thanks in advance, Ganbold ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk becomes after one month unstabled
Exact same situation with analogue phone happened to me. Apparently, there are still some serious bugs in zaptel (possibly in firmware of Digium cards, even). As far as analogue phones are concerned, my experience is that IAXy is much more reliable than TDM cards. Ronald Wiplinger wrote: Asterisk is on my box now running about one month without any troubles. Since two days I got troubles: 1. The Zapta card (2 FXS, 2 FXO) suddenly does not like one phone. It simple does not supply with a dial tone. You cannot dial. You can reach it, better say, you can dial it, it rings, but no sound. I reloaded and even restarted * without success. Than I removed the module and inserted it again, ... restarted * and it works again. 2. During a call the rtp suddenly stops, no sound in either direction. To redial from the sip phone was impossilbe, since no dial tone available. CLI did not show anything, but hitting enter gives you the next *CLI line. Reloading did not help. No dialtone on any phone, restart did help. Both situations happend within 24 hours. What could be the reason? Is there a mechanism to minimize down time? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Time sync on PRI
Hi, I am not sure about PRI, but I noticed that * does send date/time info on BRI. Perhaps installing bristuffed Asterisk would solve your problem (bristuff patches libpri, so whatever applies to BRI probably applies to PRI as well). As for syncing PC clock from ISDN line, I haven't noticed any parameter that would control something like that. It would be a nice option, but perhaps it would rise some security issues? Niksa Morten Isaksen wrote: Hi! I have this setup at a customer: PRI - (port 1) TE410P (port 2) - PABC | Asterisk Before the Asterisk part was inserted the customer claims that their PABC automatic changed the clock acourding to daylight saving time from the PRI. Now the customer says that it is not working any more. We are using pri_net signalling up against the PABC and pri_cpe on the other interface. Does Asterisk send time syncronisation on pri_net signalling? Is there a configuration setting that enables that? Is it possible to sync the computer clock with the time from the PRI? The Asterisk server is not connected to a network so NTP is not an option. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: 911 SoftHangup on SPA-3000
Well, you could use the fact that ChanIsAvail command returns session ID
together with channel name and remember it in your macro for normal
calls, something like this:
[globals]
BUSYOUTCHAN=
[macro-pstn-out]
exten = s,1,ChanIsAvail(SIP/potsoutbound)
exten = s,2,SetGlobalVar(BUSYOUTCHAN=${AVAILCHAN})
exten = s,3,Cut(FREECHAN=AVAILCHAN,,1)
exten = s,4,Dial(${FREECHAN}/${ARG1},30)
exten = s,102,Congestion
BUSYOUTCHAN variable should now contain the correct name of session to
hang up:
exten = s,x,SoftHangup(${BUSYOUTCHAN})
in your 911 macro should work. Bear in mind that this will only work if
your outbound link contains a single line. Achieving the same for
multiple lines is harder by an order of magnitude, due to limitations in
* config.
John Goerzen wrote:
On 2005-03-26, Niksa Baldun [EMAIL PROTECTED] wrote:
It is probably a SPA-3000 problem. I have tried a similar setup (not for
911, I need to hangup a phone) and it works with ISDN phones and
Swisswoice SIP phone, but not with BudgetOne, for example. The phone
just won't drop the line for some reason. Hope this helps.
It maybe set me down the right path.
As I mentioned, SoftHangup(SIP/potsoutbound) doesn't work.
However, if I get the name of a specific connection -- for instance,
SIP/potsoutbound-7d5c, I can use the Manager command Hangup to hang up
that particular connection, and it works.
However, the name of the connection isn't predictable, and I can't
figure out how to do this from the dialplan.
Suggestions are still welcome :-)
Thanks,
John
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Re: [Asterisk-Users] 911 SoftHangup on SPA-3000
It is probably a SPA-3000 problem. I have tried a similar setup (not for 911, I need to hangup a phone) and it works with ISDN phones and Swisswoice SIP phone, but not with BudgetOne, for example. The phone just won't drop the line for some reason. Hope this helps. John Goerzen wrote: Hi, I have a SPA-3000 and would like to use the 911 recipe from http://www.voip-info.org/wiki-Asterisk+tips+911. So I took the simple recipe and modified it slightly: exten = 911,1,ChanIsAvail(SIP/potsoutbound) exten = 911,2,Dial(SIP/potsoutbound/911) exten = 911,3,Hangup() exten = 911,102,SoftHangup(SIP/potsoutbound) exten = 911,103,Wait(1) exten = 911,104,Goto(1) Now, I made the appropriate changes -- Changing Zap/1 to SIP/postoutbound. Things are working fine if nobody is using the line. However, if people are using the line, ChanIsAvail doesn't detect it. I thought -- fine, make 103 be a SoftHangup so that when Dial detects the in-use condition, it will hang up. Well, asterisk -vvv shows that the SoftHangup command is running, but it's not causing the SPA-3000 to drop the line, so this creates an endless loop. I have tried the a option to SoftHangup. Any ideas? Thanks, John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create Zap channel when dialing using a bri cellular gateway
Obviously, your ISDN gateway is misconfigured somehow. I would suggest that you configure the gateway to dial some extension on your * box and see if incoming calls work. If they don't, then there is a problem with configuration of gateway's ISDN interface. If incoming calls work, then it is possible that the gateway is rejecting outgoing calls based on number called (I had that problem once), or perhaps you just forgot to pay the bill to your mobile operator :)). Niksa David Masure wrote: Hi all, I have an asterisk box set up with a bri card (using zaphfc). I have a bri cellular gateway connected to it beacuse I'd like to route all my cellular calls through that gateway. The probel I encounter is that when trying to dial a phone number, I've the message : unable to create a zap channel. My card is normally well configured because when connected to the NT, It works perfectly... The gateway is configured as a NT as well so no worry... Has anyone an idea of what I should look for ? Thank you David Masure ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Beronet BN4S0 (quad BRI) card, echo cancel, zaptel timing, bristuff ...
Bristuff works fine with Beronet cards. As far as I can see there is no difference between Beronet and Junghanns cards, not even physically. As for chan_misdn, it is still in very early stages of development, so I don't think you can expect features similar to those in bristuff, not to mention it is likely to be buggy. Robert Rozman wrote: Hi, I guess I'd need to run Beronet quad and octo bri cards under bristuff to get zaptel features (echo canceling, timing source) Am I right or could I achieve this also with chan_misdn - their native driver ? Running bristuff on Beronet cards is unsupported. Has anyone succesfully run Beronet quad BRI cards under bristuff recently ? Do they work ? Regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BRIstuff - synchronization with PSTN?
Hello, for a long time I've been having problems with analog modems and faxes communicationg over ISDN BRI lines. Now I began to suspect that this is due to * being out of sync with the PSTN. I have a quadBRI card with first two ports connected to PSTN, and defined as follows: span=1,1,0,ccs,ami span=2,2,0,ccs,ami This should mean that spans 1 and 2 are used as primary and secondary synchronization sources. However, when I check the spans with zttool it says: Sync source: Internally clocked Does this mean that ISDN interface is not synchronized with the PSTN? If so, why, and how can I correct it? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BRIstuff - synchronization with PSTN?
A bug in zttool?? OK, if that is the case, is there a way I can verify that the span is indeed synchronized with PSTN? Michael Bielicki wrote: it's a bug in zttool On Sat, 26 Feb 2005 18:40:19 +, Niksa Baldun [EMAIL PROTECTED] wrote: Hello, for a long time I've been having problems with analog modems and faxes communicationg over ISDN BRI lines. Now I began to suspect that this is due to * being out of sync with the PSTN. I have a quadBRI card with first two ports connected to PSTN, and defined as follows: span=1,1,0,ccs,ami span=2,2,0,ccs,ami This should mean that spans 1 and 2 are used as primary and secondary synchronization sources. However, when I check the spans with zttool it says: Sync source: Internally clocked Does this mean that ISDN interface is not synchronized with the PSTN? If so, why, and how can I correct it? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ericsson MD-110 and Dig-410
Your span definition should be fine (except there should be commas instead of dots, but that is probably just a typo). You need to play with various parameters on the MD-110 side, those in RODAI command, as well as SIG parameter in ROCAI command. I don't know how well is QSIG implemented in libpri, but interconnecting should be possible even without it. It is a pain in the butt, but I am afraid that trial-and-error is the only way to go. Theodoros Georgiou wrote: Hello All I am wondering is someone knows how to configure the * to work with an Ericsson MD-110 with SL60 signaling?? through a TLU76 card. What is the right configuration in the zaptel.conf ? I currently have it configured as span=3.0.0,ccs,hdb3,crc4 but it doesn't detect anything when I connect it to the PBX and no activity can be seen either in the logs or in the asterisk console. The port is responding when I connect it to the external PRI. Can anybody help ? Anyone who has seen that before ? Thanks Theo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] install BRIstuff on *@home?
Each release of BRIstuff is made for a specific * version. BRIstuff installer automatically downloads the correct version, patches and installs it. You should just run the install.sh and it will replace your current * installation. Your existing configuration (extensions.conf etc.) will not be changed. Bear in mind that you have to copy the BRI modules (qozap.ko and zaphfc.ko) manually to your /lib/modules/`uname -r`/misc directory. Erwin de Raad wrote: I'm still trying to install a HFC-s BRI card onto [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] .6 I'm new to this so I probably am overlooking the obvious. Can I just install BRIstuff onto a fresh [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] install? The BRIstuff installer downloads another * from Digium. Will this interfere with the @home install and must I comment out the Asterisk install in the BRIstuff install.sh file? Any pointers are much appreciated. Regards, Erwin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bristuff - analogue communication over ISDN
Hello, I have * config with one quadBRI card for PSTN ISDN lines and one TDM400P for analogue faxes and modems. I could never get a fax or modem to work reliably over ISDN. With bristuff 0.2.0 RC3a (* 1.0.3) modem connection would drop after a few secs, and fax would never get through if it had more than two pages. Then I upgraded to bristuff 0.2.0 RC6 (* 1.0.5) and things got better, but hardly satisfactory. Modem connection holds up to 15 min, and faxes go through although output is sometimes garbled. I would appreciate any pointer on how to begin resolving this issue. I realize that the best advice is to get rid of analogue technology alltogether, but unfortunately that is currently not an option. My zaptel.conf and zapata.conf are attached. Niksa [trunkgroups] [channels] language=en callwaiting=yes callwaitingcallerid=no threewaycalling=yes transfer=yes cancallforward=yes usecallerid=yes hidecallerid=no useincomingcalleridonzaptransfer=yes relaxdtmf=no echotraining=no echocancel=no echocancelwhenbridged=no rxgain=0.0 txgain=0.0 callgroup=0 pickupgroup=0 context=from-tcom switchtype=euroisdn pridialplan=local prilocaldialplan=local overlapdial=yes immediate=no signalling=bri_cpe group=1 channel = 1,2,4,5 context=from-mobile group=2 channel = 7,8 context=international signalling=bri_net_ptmp callreturn=yes callerid=Servis ISDN 36 group=3 channel = 10,11 signalling=fxo_ks busydetect=yes busycount=4 callerid=Fax modem 22 group=4 channel = 13 callerid=Servis analog 24 group=5 channel = 14 callerid=Dragan modem 26 group=6 channel = 15 callerid=Fax 27 group=7 channel = 16 span=1,1,3,ccs,ami span=2,2,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami bchan=1-2 dchan=3 bchan=4-5 dchan=6 bchan=7-8 dchan=9 bchan=10-11 dchan=12 fxoks=13 fxoks=14 fxoks=15 fxoks=16 loadzone=nl defaultzone=nl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Q: ISDN / E1-PRI - fax problems - Receiving and setting of Service Indicator (SIN) / Bearer Capability (BC) / High Level Compatibility (HLC) / Low Level Compatibility (LLC)
Yeah, I also noticed that * lacks the ability to forward calls based on the type of call, but I have no idea whether this issue has any priority with the development team. It is probably better to ask this question on Asterisk-Dev mailing list. Frank Sautter wrote: hi, i have the problem that i'm not able to set and receive the Service Indication (SIN) from our E1-PRI and from our ericsson BP250. The problem is, that the Bearer Capability (BC) together with the High Level Compatibility (HLC) and Low Level Compatibility (LLC) forms the Service Indicator (SIN). The SIN is used to determine if the call is voice, fax or data. It's essential to set the SIN so the called party is able which device has to answer a call (e.g. telephone or fax) as far as i dug into the source neither the BC nor the HLC or LLC data is forwarded to a dialplan variable and only the BC is decoded in libpri. has anyone a solution for this? is there any usable documentation on the HLC or LLC octets (bytes)? i searched etsi and was overwhelmed with the searchresults (1531). what i need to modify libpri would be a table of possible values and where to find the HLC and LLC fields in the D-Channel. regards frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 lost after reboot
I had exactly the same issue with the newest card I got. I tried it with Zaptel drivers from CVS HEAD and the problem disappeared. It could be that older drivers don't work with the latest cards. Mark wrote: Do you have your zaptel drivers set to start when the system is rebooted? If not, try rebooting and issue the modprobe zaptel and modprobe wctdm commands to manually start them. You could also issue the lsmod command after a reboot to see if zaptel and wctdm are running. I had problems with the zaptel startup script, but for whatever reason it works now. Good luck! Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, January 16, 2005 7:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] TDM400 lost after reboot Hi My card is working, but when I reboot the machine, most of the times it is not working, I get ztcfg: ZT_CHANCONFIG failed on channel 1: No such device or address (6) To make it work again I have to shut down, remove the card, reboot so kudzu will remove the config. shut down again, put the card back in, reboot, now kudzu see it, I choose Ignore and then it's working again (until the next reboot). I'm on WBEL 3.0 and the card is not sharing is IRQ. Is anybody else having this problem ? When kudzu see it (as a Jens Schoenfeld Intel 537), what should I choose ? Is there something I can do to prevent this from happening ? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Environment variables
Is accessing Linux environment variables available in Asterisk stable? I seem to cannot be able to do it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SCCP questions
Hello,
I am a Kirk IP600 user too, and I had partial success in getting it to
work with chan_sccp. I changed the line 133 in chan_sccp.c to the
following:
if ( (!s-device) (mid != RegisterMessage mid != AlarmMessage
mid != KeepAliveMessage mid != IpPortMessage)) {
And then I was able to register IP600 hansets, but only to find there
are other problems. I am currently communicating with Kirk support in
order to see what is happening. When I have more info I will send it to
you.
Niksa
Julien Goodwin wrote:
On Thu, Jan 13, 2005 at 01:09:54PM +0200, Kelemen Zoltan arranged a set of bits into the following:
Hi!
I have two, not too related questions:
- the probably simpler one: if anyone can help me out using a Cisco
7905G with chan_sccp? I did already managed to get it working with a SIP
image, I'd just like to see it work with this one as well. It's probably
something I screw up with the configuration, as the phone registers,
only I don't get any lines with it, although I have it configured it to
auto-login.
excerpt from my sccp.conf
-snip--
[SEP001193C2ABFC]
device=SEP001193C2ABFC
type=7905
autologin= c7905
callerid="cisco 7905"
[c7905]
line = 1045
-snip--
(I have a default extension set for the entire sccp.conf, so that
shouldnt(?) be the issue)
That's the contents of a skinny config file, the two have different
formats. Here's what I use for one of my phones:
---snip---
[SEP003080628DD7]
type= 12
autologin = phone4
description = Cisco Phone 4
[phone4]
id = 4005
pin = 1237
label = Phone4
description = Phone4
callwaiting = 0
mailbox = 4005
cid_num = 4005
cid_name= Cisco Phone 4
--- endsnip ---
And I call with Dial(SCCP/phone4)
I have the XMLDefaultConf in place, tftp server running, although that's
about it. I would appreciate any pointers in this general direction.
What am I missing? :)
See above, copy the format, just use your own data.
The second, much more thorny question is: did anyone had any success on
using a KIRK IP600 with asterisk?
- The only thing I really found on the net were a couple of emails on
this list, that didn't get me too far.
The KIRK IP600 is a DECT (cordless) to IP solution, with support for
SCCP and H323. The SCCP interface was designed specifically to be
interoperable with Cisco Call Manager, and it emulates a 7940 for each
of the phones it has registered.
With chan_skinny I managed to register the phones, they've got tone, but
they would not ring out.
With chan_sccp I had no luck at all, I'm getting the following messages
on the CLI:
== Got message AlarmMessage
== Got message RegisterMessage
== Sending Packet Type RegisterRejectMessage (37 bytes)
Note: the two modules were NOT tried at once.
So far I didn't have time to check it out with h323, but If anyone had
it working that way, I'm interested in that one as well.
H323 is probably the one to try, but fix your sccp config like the above
(a compile fix for asterisk CVS has just been committed) and let me know
what you get (in debug mode very verbose). If anyone in Australia has one
or is able to arrange a loan for a few days I'd probably be able to make
them work, but without being able to hack on the code with the device it
makes it quite hard.
I'm just writing some code to see if I can fix the "client sent
IPPortMessage without first registering" that someone got before, if
anyone is able to duplicate that drop me a line and I'll see if my patch
works against it.
Thanks,
Julien Goodwin
chan_sccp developer
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Re: [Asterisk-Users] Build PWLIB
You need the following to make OH323 work:
PWLIBDIR=/usr/local/src/pwlib
export PWLIBDIR
OPENH323DIR=/usr/local/src/openh323
export OPENH323DIR
LD_LIBRARY_PATH=${LD_LIBRARY_PATH}:${PWLIBDIR}/lib:/usr/local/lib:${OPENH323DIR}/lib
export LD_LIBRARY_PATH
Replace '/usr/local/src/' with whatever directory you put your sources
in. Make sure you put these in one of your startup scripts, so they are
executed each time the system is run.
Walid Azab wrote:
I
am trying to build PWLIB to get OH323 up and running.
I
am not an expert in linux but can someone help telling me how I can do
the following:
How
can I add a directory to LD_LIBRARY_PATH?!
Thanks
in advance
--
For unix.
--
1. If you have not put pwlib it
into your home directory (~/pwlib) then
you will have to defined the environment variable PWLIBDIR to
point to
the correct directory.
Also make sure you have added the $PWLIBDIR/lib directory to
your
LD_LIBRARY_PATH environment variable if you intend to use shared
libraries (the default).
-
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Re: [Asterisk-Users] So many Asterisk Patches - Which do I choose and use?
There is no easy answer to your question. If you ask me, I prefer not to use any patches, except that I am forced to use bristuff because I have quadBRI ISDN cards. Bristuff patches Zaptel in order to enable using quadBRI and octoBRI cards, and also adds some features to *. More info on www.junghanns.net. Like you said, really valuable patches will make it to the CVS sooner or later, so I prefer to wait because it makes installation and maintenance easier. I use Gentoo with 2.6 kernel. I am not sure whether you will get any benefits from upgrading, but I didn't have any problems with it (except that I had to migrate from devfs to udev, but that issue exists with 2.4 kernel too). Paul Rodan wrote: Ok, I usually use the latest stable CVS, with no patches or modifications. If figured if there was a worthwhile patch, Mark would have already included it. However, there was that neat patch about being able to press a certain key and itd begin recording in mid-stream, that was an awesome feature and I patched my latest features.c file with that patch. But I keep seeing mentions of other patches, specifically something about the MOH patch, the BRISTUFFED patch, and now Im hearing about a Super Parking Lot patch? For now Ive been using the mpg123 method, it tends to work for me, but if I can save CPU/RAM and other troubles by using another format, which one do I go with? What is BRISTUFFED? And if Im right, the super parking lot patch allows for call parking based on context, a way to break it apart, instead of making it universal across the whole system (where can I find this patch)? So Im going to ask the question, if I were to install the latest CVS Stable tonight, which patches should I install on it before compiling? Also, Im using Gentoo Linux, with the 2.4.26-r9 gentoo kernel. Ive seen issues with people making Asterisk work perfectly with the 2.6 kernel so Ive stayed clear of it, but I still see people fighting to make it work and such, I saw one post a while back about the benefits using Asterisk w/ the 2.6 kernel, can somebody please refresh my memory? What are the benefits of using Asterisk with the 2.6 kernel? Im trying to get the most out of my system. Any help in making tonights compile/upgrade go perfect would be greatly appreciated. Thanks, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to buy a quadBRI?
You can try www.beronet.com, they sell entire Digium product line + quad/octoBRI. Johannes Morgenroth wrote: Who is selling the quadBRI cards, which is everybody talking about? And how much does is costs? I'm looked on the site of junghanns.net, but there is no selling information. A seller in germany is favored. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk variables - size limitation?
Does anybody know if there is any size limitation on * variable, or is it only limited by available memory? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analogue RAS Server
I don't think it's possible. Asterisk would have to emulate analog modem, and I believe that feature is not (at least yet) implemented. Daniel Niasoff wrote: Hi, Does anyone have any idea how to set up Asterisk so that it can act as an Analogue Remote Access Server. Ive looked around and as far as I can see it will only act as an ISDN Ras server. Thanks Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Execute dialplan command at startup
I tried it, and it works, but it is hardly an ideal solution. If you, for example, forget yourself and execute 'restart now' from * console, .call file won't be generated and you may spend hours wondering what went wrong. I believe the lack of some kind of 'autostart' context is a major flaw in design, but then again, I guess it could easily be implemented. Peer Oliver Schmidt wrote: Bill Seddon wrote: How can Asterisk be configured to execute some number of dialplan commands when it is started or restarted? [..] In the meantime I'm hoping that it is possible to use the built-in database and be able to run some kind of autostart context. Does such a facility exist? Without getting into details, I would create a call file in the outgoing spool directory of asterisk within the asterisk startup script which calls a specific application. Haven't tried, but should easily work. Let me know, how it works. rgds pos ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wait indefinitely?
Hello, is it possible to wait indefinitely (i.e. until user hangs up) somewhere in the dialplan? I tried Wait(-1), but it doesn't work. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk startup
Is there a way to execute specific applications at Asterisk startup (like startup context or something)? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZapRAS with BRI
Hello, I am using * with latest bristuff. I tried to make use of ZapRAS application (with the little or no documentation available), but I get the following error when trying to establish RAS connection: app_zapras.c:149 run_ras: wait4 returned -1: No child processes Anyone has an idea what this means? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZapRAS with BRI
Thanks, that was it. Now I can accept incoming RAS connections. But, what about outgoing connections? Is it possible to make * behave like ISDN router, i.e. have it establish a connection when someone on the LAN requests specific IP address? Thanks again. Steven Critchfield wrote: On Thu, 2005-01-06 at 18:10 +, Niksa Baldun wrote: Hello, I am using * with latest bristuff. I tried to make use of ZapRAS application (with the little or no documentation available), but I get the following error when trying to establish RAS connection: app_zapras.c:149 run_ras: wait4 returned -1: No child processes Anyone has an idea what this means? My guess is that you are having trouble with the PPP daemon. If you patched and installed your own PPPD like I remember the limited docs saying to do, it may not be in /usr/sbin/pppd like is compiled into the app. It is likely it is in /usr/local/sbin/pppd. Just to help out also, here was the line we used when testing and actually brought it up with an ISDN Ascend Pipeline router calling into asterisk. ;exten = 9022,1,ZapRas(debug|64000|noauth|netmask|255.255.255.0|192.168.10.1:192.168.10.2) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZapRAS with BRI
I figured it would be something like that. Well, at least I have something to work on. Thanks again. Steven Critchfield wrote: On Thu, 2005-01-06 at 22:05 +, Niksa Baldun wrote: Thanks, that was it. Now I can accept incoming RAS connections. But, what about outgoing connections? Is it possible to make * behave like ISDN router, i.e. have it establish a connection when someone on the LAN requests specific IP address? This would be a bit harder. Start with detecting when the route is down and the specific IP needs to come up. If you can detect that, then you just need to drop an appropriate .call file and it will dial out and connect. I'm assuming this is all in an attempt to do dial on demand out to remote locations. I'm not sure how to do the detect you need to route so fire a script. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ericsson 4422/4425 phones
Hello, anybody managed to get Ericsson Dialog 4422 or 4425 IP phone to work with Asterisk/OH323/gnugk? They register with the gatekeeper, but that is about it. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Line-in as MOH source
Hello, Most traditional PBX-es have the ability to use external audio source (e.g. radio tuner) for music on hold. This is also useful because you can let your users listen to radio by dialing some extension. I wanted to achieve the same on asterisk, and chan_alsa seemed the logical choice. I installed ALSA drivers, connected the radio to line-in and added the folowing to extensions.conf: exten = *55,1,Dial(Console/Line) And indeed, now I could listen to radio by dialing *55. There are some problems, however: 1. Asterisk treats this as a normal call, so only one user can listen at a time. Is there a way to let several user listen simultaneously? 2. As this is a low priority call, it should be dropped when incoming call is sent to a channel which dialed it. I guess SoftHangup could be used, but I don't know how to determine which channel to hangup. 3. I have no idea how to use this as a source for music on hold. Any help would be appreciated. Niksa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
