Re: [asterisk-users] pri call by call trunking?
On 8/2/07, Don Kelly [EMAIL PROTECTED] wrote: Hi, Erik, Never heard of call-by-call trunking. Are you in Minnesota? What carrier are you using? Yes I am...this is for one of our branch offices, though, outside of Boston, MA. -Erik ___ --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] pri call by call trunking?
We spent a considerable amount of time getting an A101 up and running. Try to find out what type of switch you are connecting to. In our case, we were working against a Nortel. For some reason, if we used ni2, it would not work. Finally setting the switchtype to 5ess or DMS100 would work and now everything sings. Hope that helps. Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Wednesday, August 01, 2007 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] pri call by call trunking? Call Sangoma On 8/1/07, Erik Anderson [EMAIL PROTECTED] wrote: On 8/1/07, John covici [EMAIL PROTECTED] wrote: I had some troubles -- try setting the timing parameter to 0 (second one in your span) and see if that helps. If I'm reading the docs correctly, this param should only be set to 0 if you *never* want to use the T1 connected to this port for timing. That's not the case in my situation, as I need to be syncing with the telco's clock. That said, in the interest of troubleshooting, I did try setting it to zero - this didn't fix the problem. -erik ___ --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] pri call by call trunking?
Hi, Erik, Never heard of call-by-call trunking. Are you in Minnesota? What carrier are you using? --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Anderson Sent: Wednesday, August 01, 2007 12:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] pri call by call trunking? I've been working with a telco for the past two days trying to get a PRI span up and running. This is a small-ish telco and I get the feeling they don't do this very often. Anyway, they specified a pretty standard setup: ni2 switchtype, esf framing, b8zs coding, etc. All of my b-channels are up, but we're having a heck of a time getting the d-channel to come up. He finds out that this is an asterisk system and says that to get this working, I'm going to need to turn on call-by-call trunking. Have any of you heard of this? I certainly haven't. A quick google search doesn't turn up anything. Thoughts? This is a Sangoma A102 card, by the way. In this case, though, I don't think that's of any relevance. -Erik -- Erik Anderson http://andersonfam.org ___ --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] pri call by call trunking?
On 8/1/07, Erik Anderson [EMAIL PROTECTED] wrote: I've been working with a telco for the past two days trying to get a PRI span up and running. This is a small-ish telco and I get the feeling they don't do this very often. Anyway, they specified a pretty standard setup: ni2 switchtype, esf framing, b8zs coding, etc. All of my b-channels are up, but we're having a heck of a time getting the d-channel to come up. He finds out that this is an asterisk system and says that to get this working, I'm going to need to turn on call-by-call trunking. Have any of you heard of this? I certainly haven't. A quick google search doesn't turn up anything. Thoughts? This is a Sangoma A102 card, by the way. In this case, though, I don't think that's of any relevance. Yes it is. Try setting TDMV_DCHAN = 0 in your wanpipe1.conf ___ --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] pri call by call trunking?
I've been working with a telco for the past two days trying to get a PRI span up and running. This is a small-ish telco and I get the feeling they don't do this very often. Anyway, they specified a pretty standard setup: ni2 switchtype, esf framing, b8zs coding, etc. All of my b-channels are up, but we're having a heck of a time getting the d-channel to come up. He finds out that this is an asterisk system and says that to get this working, I'm going to need to turn on call-by-call trunking. Have any of you heard of this? I certainly haven't. A quick google search doesn't turn up anything. Thoughts? This is a Sangoma A102 card, by the way. In this case, though, I don't think that's of any relevance. -Erik -- Erik Anderson http://andersonfam.org ___ --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] pri call by call trunking?
what channel are they putting the Dchannel on? Post your zapata.conf and zaptel.conf On 8/1/07, Erik Anderson [EMAIL PROTECTED] wrote: I've been working with a telco for the past two days trying to get a PRI span up and running. This is a small-ish telco and I get the feeling they don't do this very often. Anyway, they specified a pretty standard setup: ni2 switchtype, esf framing, b8zs coding, etc. All of my b-channels are up, but we're having a heck of a time getting the d-channel to come up. He finds out that this is an asterisk system and says that to get this working, I'm going to need to turn on call-by-call trunking. Have any of you heard of this? I certainly haven't. A quick google search doesn't turn up anything. Thoughts? This is a Sangoma A102 card, by the way. In this case, though, I don't think that's of any relevance. -Erik -- Erik Anderson http://andersonfam.org ___ --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] pri call by call trunking?
On 8/1/07, C F [EMAIL PROTECTED] wrote: what channel are they putting the Dchannel on? Post your zapata.conf and zaptel.conf The D channel is on 24. zaptel.conf: loadzone=us defaultzone=us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 zapata.conf lpdlnx04 asterisk # cat zapata.conf ;autogenerated by /usr/local/sbin/config-zaptel do not hand edit ;Zaptel Channels Configurations (zapata.conf) ; ;For detailed zapata options, view /etc/asterisk/zapata.conf.orig [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A102 port 1 [slot:10 bus:2 span: 1] switchtype=national context=from-pstn group=1 signalling=pri_cpe channel = 1-23 ___ --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] pri call by call trunking?
On 8/1/07, John covici [EMAIL PROTECTED] wrote: I had some troubles -- try setting the timing parameter to 0 (second one in your span) and see if that helps. If I'm reading the docs correctly, this param should only be set to 0 if you *never* want to use the T1 connected to this port for timing. That's not the case in my situation, as I need to be syncing with the telco's clock. That said, in the interest of troubleshooting, I did try setting it to zero - this didn't fix the problem. -erik ___ --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] pri call by call trunking?
I had some troubles -- try setting the timing parameter to 0 (second one in your span) and see if that helps. on Wednesday 08/01/2007 Erik Anderson([EMAIL PROTECTED]) wrote On 8/1/07, C F [EMAIL PROTECTED] wrote: what channel are they putting the Dchannel on? Post your zapata.conf and zaptel.conf The D channel is on 24. zaptel.conf: loadzone=us defaultzone=us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 zapata.conf lpdlnx04 asterisk # cat zapata.conf ;autogenerated by /usr/local/sbin/config-zaptel do not hand edit ;Zaptel Channels Configurations (zapata.conf) ; ;For detailed zapata options, view /etc/asterisk/zapata.conf.orig [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A102 port 1 [slot:10 bus:2 span: 1] switchtype=national context=from-pstn group=1 signalling=pri_cpe channel = 1-23 ___ --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 [EMAIL PROTECTED] ___ --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] pri call by call trunking?
Call Sangoma On 8/1/07, Erik Anderson [EMAIL PROTECTED] wrote: On 8/1/07, John covici [EMAIL PROTECTED] wrote: I had some troubles -- try setting the timing parameter to 0 (second one in your span) and see if that helps. If I'm reading the docs correctly, this param should only be set to 0 if you *never* want to use the T1 connected to this port for timing. That's not the case in my situation, as I need to be syncing with the telco's clock. That said, in the interest of troubleshooting, I did try setting it to zero - this didn't fix the problem. -erik ___ --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
