I have just started developing asterisk, and am trying to start simple. I have a X100P device and an S100U device. I am trying to use the examples provided, where I add a few lines to the /etc/zaptel.conf, /etc/asterisk/zapata.conf, and /etc/asterisk/extensions.conf so that I may connect an analog line to the X100P and an analog phone to the S100U. When I dial the analog line, it should ring the phone. My problem is that when I connect the analog line to the X100P, I receive a busy tone when dialing it. I have switched tip and ring and still have the same result. Also, I receive no dialtone from the S100U. When I run asterisk, I receive two warning messages, both have number 8192 associated with them (not sure if that's significant). The first one reads: File chan_iax2.c, line 4952 (set_config): ignoring ports for now. ==Registered channels type 'IAX2' ==Using TOS bits 16 ==IAX ready and listening on 0.0.0.0 port 4569 The second one reads: File chan_oss.c line 419 (soundcard_init): unable to open /dev/dsp: no such device ==No sound card detected - - console channel will be unavailable ==Turn off OSS support by adding 'noload=chan_oss.so' Asterisk still runs and recognizes both channels when I run command "zap show channel 1" and "zap show channel 2", even with these warnings. I have also went into /etc/asterisk/modules.conf and added the lines noload=chan_iax2.so and noload=chan_oss.so, which took care of the warnings, but I'm not sure if that will introduce problems later on. I am wondering if there is some sort of initialization that needs to be done with the Zaptel devices? The steps that I followed were off the * support site, under the FAQ "I just received my DevKit Lite. What should I do now, and can you please show me." Thanks in advance, Jesse
On Thu, 2003-04-17 at 08:42, JKNUTSEN@UP.COM wrote:> I have just started developing asterisk, and am trying to start simple. I > have a X100P device and an S100U device. I am trying to use the examples > provided, where I add a few lines to the /etc/zaptel.conf, > /etc/asterisk/zapata.conf, and /etc/asterisk/extensions.conf so that I may > connect an analog line to the X100P and an analog phone to the S100U. When > I dial the analog line, it should ring the phone. My problem is that when > I connect the analog line to the X100P, I receive a busy tone when dialing > it. I have switched tip and ring and still have the same result. Also, I > receive no dialtone from the S100U. > > When I run asterisk, I receive two warning messages, both have number 8192 > associated with them (not sure if that's significant). The first one > reads: File chan_iax2.c, line 4952 (set_config): ignoring ports for now. > ==Registered channels type 'IAX2' > ==Using TOS bits 16 > ==IAX ready and listening on 0.0.0.0 port 4569 > > The second one reads: File chan_oss.c line 419 (soundcard_init): unable > to open /dev/dsp: no such device > ==No sound card detected - - console channel will be unavailable > ==Turn off OSS support by adding 'noload=chan_oss.so' > > Asterisk still runs and recognizes both channels when I run command "zap > show channel 1" and "zap show channel 2", even with these warnings. I have > also went into /etc/asterisk/modules.conf and added the lines > noload=chan_iax2.so and noload=chan_oss.so, which took care of the > warnings, but I'm not sure if that will introduce problems later on. > > I am wondering if there is some sort of initialization that needs to be > done with the Zaptel devices? The steps that I followed were off the * > support site, under the FAQ "I just received my DevKit Lite. What should I > do now, and can you please show me."I'll answer the easy parts for you. Unless you wish to use a soundcard as a console device on asterisk, not loading the chan_oss will be fine. As for IAX2, unless you are connecting to another asterisk box with IAX2, it is okay to not load it as well. Just try to remember to look there if you ever do want to connect multiple asterisk boxes together. Now for the harder part. When you call you asterisk machine, do you see any indications in the asterisk console that it received a call? I ask this since asterisk could return a busy tone to you if it was unable to complete the call to the S100U. Have you tried getting the the call into the X100P to go to some voice prompts to verify this one piece. -- Steven Critchfield <critch@basesys.com>
First off, thanks for the answers to the warning messages. As far as the harder part, I'm not seeing any indications in the asterisk console that I'm receiving a call. I tend to believe that my problem is not with asterisk, but rather with the boards. If I call the analog line when asterisk is not running I still receive a busy signal, but as soon as I unplug the line from the board I get ring no answer. So it seems to me that something on the board is causing the line to think its going off hook. If it is with the board, this may not be the right place to post, but I thought people might be familiar with the Wildcard boards here. Again, thanks in advance, Jesse On Thu, 2003-04-17 at 08:42, JKNUTSEN@UP.COM wrote:> I have just started developing asterisk, and am trying to start simple.I> have a X100P device and an S100U device. I am trying to use the examples > provided, where I add a few lines to the /etc/zaptel.conf, > /etc/asterisk/zapata.conf, and /etc/asterisk/extensions.conf so that Imay> connect an analog line to the X100P and an analog phone to the S100U.When> I dial the analog line, it should ring the phone. My problem is thatwhen> I connect the analog line to the X100P, I receive a busy tone whendialing> it. I have switched tip and ring and still have the same result. Also,I> receive no dialtone from the S100U. > > When I run asterisk, I receive two warning messages, both have number8192> associated with them (not sure if that's significant). The first one > reads: File chan_iax2.c, line 4952 (set_config): ignoring ports for now. > ==Registered channels type 'IAX2' > ==Using TOS bits 16 > ==IAX ready and listening on 0.0.0.0 port 4569 > > The second one reads: File chan_oss.c line 419 (soundcard_init): unable > to open /dev/dsp: no such device > ==No sound card detected - - console channel will be unavailable > ==Turn off OSS support by adding 'noload=chan_oss.so' > > Asterisk still runs and recognizes both channels when I run command "zap > show channel 1" and "zap show channel 2", even with these warnings. Ihave> also went into /etc/asterisk/modules.conf and added the lines > noload=chan_iax2.so and noload=chan_oss.so, which took care of the > warnings, but I'm not sure if that will introduce problems later on. > > I am wondering if there is some sort of initialization that needs to be > done with the Zaptel devices? The steps that I followed were off the * > support site, under the FAQ "I just received my DevKit Lite. What shouldI> do now, and can you please show me."I'll answer the easy parts for you. Unless you wish to use a soundcard as a console device on asterisk, not loading the chan_oss will be fine. As for IAX2, unless you are connecting to another asterisk box with IAX2, it is okay to not load it as well. Just try to remember to look there if you ever do want to connect multiple asterisk boxes together. Now for the harder part. When you call you asterisk machine, do you see any indications in the asterisk console that it received a call? I ask this since asterisk could return a busy tone to you if it was unable to complete the call to the S100U. Have you tried getting the the call into the X100P to go to some voice prompts to verify this one piece. -- Steven Critchfield <critch@basesys.com> _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
> Hi, > > I have a question regarding X100P card. > > I have one X100P card in an * box. > I have the telco line connected to the line port of the X100P card, and an > analog phone connected to the phone port of the X100P card. > > My question is: > How to make ringing the analog phone connected to the phone port when you > receive a VoIP call? > > Thanks. > > GIBERT Fr?d?ric > Mobile: +33 6 72 08 35 16 > Fax : +33 1 30 71 39 33 > Mail : frederic.gibert@viginetworks.fr > > Bureau Paris : > Ste VIGINETWORKS (Chez CAP retraite) > 137, rue vielle du temple > 75003 Paris > France > >-------------- next part -------------- A non-text attachment was scrubbed... Name: winmail.dat Type: application/ms-tnef Size: 2020 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040901/d3dcf9ac/winmail.bin
GIBERT Fr?d?ric wrote:>I have one X100P card in an * box. >I have the telco line connected to the line port of the X100P card, and an >analog phone connected to the phone port of the X100P card. > >My question is: >How to make ringing the analog phone connected to the phone port when you >receive a VoIP call? > >The phone port on the X100P is only for use when the system is powered down. For example, during a power outage the connected phone will ring and you can still answer and place calls with the connected phone.
Gilbert, The phone port is only a loop thru port for the analogue line. It is not an FXS port. Dave> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] > Sent: 01 September 2004 09:32 > To: Asterisk-Users; Asterisk-Dev-Admin > Subject: [Asterisk-Users] X100P question > > Hi, > > I have a question regarding X100P card. > > I have one X100P card in an * box. > I have the telco line connected to the line port of the X100P card, and an > analog phone connected to the phone port of the X100P card. > > My question is: > How to make ringing the analog phone connected to the phone port when you > receive a VoIP call? > > Thanks. > > GIBERT Fr?d?ric > Mobile: +33 6 72 08 35 16 > Fax : +33 1 30 71 39 33 > Mail : frederic.gibert@viginetworks.fr > > Bureau Paris : > Ste VIGINETWORKS (Chez CAP retraite) > 137, rue vielle du temple > 75003 Paris > France > > << File: ATT00015.txt >>-------------- next part -------------- A non-text attachment was scrubbed... Name: winmail.dat Type: application/ms-tnef Size: 2364 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040901/bb15a58f/winmail.bin