Re: [Asterisk-Users] ISDN Hardware
Have a look at http://www.sirrix.de/content/pages/pci4s0.htm Currently they don't have an english website but in short this card offers 4 Ports (8 Channels) and you are able to define the mode for each channel (PtmP, PtP, TE, NT) on your own. They have asterisk-drivers available (for Kernel 2.4) and what is very interesting about this card is, that if you have ISDN-Telephones in your environment you can connect them not only ti the astersik but when there is a connecten ISDN (internal) -- asterisk -- ISDN (external) then this will happen without asterisk beeing directly involved because these calls are connected through on the hardware without any echo-problems from transcoding the audio. All asterisk features are usable with this card including Fax based on spanDSP. The price is at around EUR 500,- Jui Jeff Lists wrote: I have a test install set up as follows Grandstream 102 Asterisk X100P Adtran Express 3000 --- ISDN line to PSTN Most things I want to do work fine except I do have some intermittent problems with an echo. I am assuming that this is created in the link between the X100P and the Adtran terminal adaptor. I am not sure of the proper terms but on the PSTN side of the Adtran there is one pair of wires which carry two channels. On the interior side of the Adtran there are two pairs of wires which each carry one channel (is this a BRI channel??). In the final install I will be replacing the X100P with some type of ISDN card. I currently have three ISDN pairs which create six lines. I would like to get enough hardware to be able to add one more pair of lines soon. My question is what type of ISDN card do I need? Will this replace the Adtran or plug into it? Since this is all digital is it a safe assumption that my echo problems will be gone? Thanks, Jeff ___ 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] 1.0.2-BRIstuffed-0.2.0-RC2b and '*8' calls dropping
Hi KPJ, btw, there is a problem with make loadNT (zaphfc) and Kernel 2.4 systems that should be fixed. I hope you already know about this IRQ_NONE issue! the problem is with line 578 in zaphfc.c saturn:/usr/src/bristuff-0.2.0-RC5/zaphfc # make loadNT cc -c zaphfc.c -D__KERNEL__ -DMODULE -DEXPORT_SYMTAB -fomit-frame-pointer -O2 -Wall -I/usr/src/linux/include -Wall -DMODVERSIONS -include /usr/src/linux/include/linux/modversions.h -DCONFIG_ZAPATA_BRI_DCHANS zaphfc.c: In function `hfc_interrupt': zaphfc.c:578: error: `IRQ_NONE' undeclared (first use in this function) zaphfc.c:578: error: (Each undeclared identifier is reported only once zaphfc.c:578: error: for each function it appears in.) zaphfc.c:578: Warnung: `return' with a value, in function returning void zaphfc.c: In function `hfc_findCards': zaphfc.c:939: Warnung: int format, long unsigned int arg (arg 8) make: *** [zaphfc.o] Fehler 1 best regards Jui Klaus-Peter Junghanns wrote: Hi Mark, please take a look at bristuff 0.2.0-RC5 which uses * 1.0.5: http://www.junghanns.net/asterisk/downloads/bristuff-0.2.0-RC5.tar.gz best regards Klaus Am Freitag, den 28.01.2005, 14:35 +0200 schrieb Mark Elkins: I'm using Asterisk 1.0.2-BRIstuffed-0.2.0-RC2b - when anyone picks up a call with '*8' - the call will drop after about 20 or so seconds. Is this a general problem with Asterisk 1.0.2? As this is the latest release that it appears Klaus-Peter Junghanns has for public consumption - is there anything I can patch for just this problem - or has Klaus-Peter Junghanns (or anyone else) been quietly busy with a BRIstuffed patch that works against Asterisk Head? I also notice that I can't seem to re-compile the H323 stuff any more... with this release... ___ 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] SoftClient for Pocket PC
http://www.xten.com/index.php?menu=productssmenu=xproppc Jui richard Coco wrote: Hi List, Is it possible to install a soft client on my Pocket Loox 610 (F.C.Siemens) an register it with asterisk? any suggestions? thx in advance. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ 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] ISDN-Phone (HFC) =*=SIP-Provider: audio only in one direction, no nat problem
Hi List! I have an interesting problem. I am behind a NAT Firewall which works fine with SIP. I am connected to T-DSL in Germany and there the DSL-Connection is interrupted every 24hours and buck a few seconds later with a new dynamic IP. My Asterisk is registered with several SIP-Providers and this works fine. In addition the *-Server has a HFC-S ISDN interface card installed in NT mode (using zaphfc) with an ISDN-Phone connected. Everything works fine when I make a call from the ISDN-Phone through my SIP-Provider to other Phones or SIP-Users (external). BUT: As soon as I get the new IP (either automatically due to the forced interrupt of my DSL line each 24hrs, or manually forced) every still seems to work fine but audio goes only in one direction from now on (alwasy I can't hear the other party but they can hear me and signalling also works fine). So I make a call, but while the phone I am calling rings I can't hear the ringtone in my phone. When the other side answers the phone I can't hear them while they can hear me loud and clear. A relaod on the CLI solves the problem till next IP-Change. I know there were already some things reported with dynamic IP's but in most cases nothing worked anymore after the IP changed. What can I do (maybe with the settings ind some conf-files). In addition I found that if I have srvlookup=yes in my sip.conf Asterisk can't register with my sip provider. But many example configs dsay you should use srvlookup=yes and I hoped that this might solve my problem but I can't use this setting set to yes at all. Any ideas on what to try? Thanks, Jui ___ 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] Echo Training - how long
Does echotraining also work with HFC-S controllers or is this feature restricted to some digium controllers? So does it make sense at all to switch it on for HFC-S controller used in NT-Mode with ISDN-Phones connected? Jui Rich Adamson wrote: I have echo training set on in my zapata.conf file for a X101P card: echocancel = yes echocancelwhenbridged = yes echotraining = yes Now, I know that echo cancellation is a black art, but I am finding that at the beginning of a call bridged between a SIP channel and a Zap channel the voice quality is poor to abysmal for the first few seconds, but as the call progresses, esp after about 30 seconds, the call quality becomes very acceptable. Should echo training take that long? Is it, in fact, echo training or some thing else? Has any one got any guidance on ET other than what is in the wiki, which I find to be very hard to follow? Try echotraining=800 and see if that makes a difference. ___ 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] Problems with a combination of AVM B1 and HFC-S on Kernel 2.6.x
Hello List! Several users of a german VoIP-Forum experience the following similar problems when using CAPI with an active ISDN-Card AVM B1 PCI connected to the PSTN and a HFC-S in NT-Mode used as an internal S0-Bus to connect ISDN-Phones. The problem is, that when making a phone call from an ISDN-Phone connected to the HFC-S-Card that is meant to be routed to the PSTN through the AVM B1, the voice quality is very bad. It sounds as if someone has a really bad, bad cold. What makes us wander is, that if a call is made from a SIP-client through the AVM B1 to the PSTN, everything ist fine. Calls between SIP-clients and ISDN-Phhones connected to the HFC-CArd are also ok. In addition one user found that if one monitors/records a phonecall made from an ISDN-Phone to the PSTN through the AVM B1, the recorded sound is also fine, only the live sound is that bad and we have no more ideas what to try to figure out where the problem is. What we heard, but we still have to verify this is, that on a Linx with Kernel 2.4 everything seems to work. But this is still unverified. The Kernel is 2.6.8 in my case the actual updated Kernel of a SuSE 9.2 Distribution. We were also testing different Motherboards and made sure that there are no IRQ-Conflicts between HFC and AVM card, etc. The Test-Systems had no load and are running smoothly. Any ideas? Thanks, Jui ___ 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
