Hi, Thanks for the reply. I tried changing my allow and disallow entries to match yours below but still no luck. Could my problems be bandwidth related? If so what amount of bandwidth should I request?
Cheers Edward -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff Glassman Sent: Sunday, March 13, 2005 12:17 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RE: Asterisk-Users Digest, Vol 8, Issue 88 These allow and disallow work with NuFone for me disallow=all allow=ulaw allow=alaw allow=gsm Jeff Message: 11 Date: Fri, 11 Mar 2005 11:15:51 +0100 From: "Edward Banfa" <[EMAIL PROTECTED]> Subject: [Asterisk-Users] NuFone Configuration [problem] To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii" Hello, I am trying to configure the my asterisk box here with the following **iax.conf*** [NuFone] type=peer host=switch-1.nufone.net secret=xxxxxx ***extensions.conf:*** exten => _1NXXNXXXXXX,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN} exten => _011N.,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN} I have a couple of Xlite softphones and 2 analogue phones connected to a mediatrix 1102 connected to our lan. The mediatrix talks sip to the asterisk box on the lan. We are running asterisk on FC3 . SOFTPHONES[XLITE] ---SIP--> ASTERISK----IAX--->NUFONE[ASTERISK] ANALOGPHONES---MEDIATRIX_1102---SIP--->ASTERISK---IAX--->NUFONE[ASTERISK ] Well the problem goes something like this. 1) I can dial a number form the softphones and when the call is answered I can hear the user on the other end but the user can't hear me 2) I can dial a number from the analog phones (via mediatrix tru to asterisk)(the mediatrix is properly registered with our asterisk box) and when the call is answered both ends can't hear a word, its just silent. I think I am having a codec problem here. What am I doing wrong. We would sincerely appreciate any help/pointers. Thank you all Edward Banfa ******EXTENSION.CONF******* [general] static=yes [from-sip] exten => 100,1,Dial(SIP/edward,20) exten => 100,2,Hangup exten => 101,1,Dial(SIP/phone1,20) exten => 101,2,Hangup exten => 102,1,Dial(SIP/phone2,20) exten => 102,2,Hangup exten => _1NXXNXXXXXX,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN} exten => _011N.,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN} *****IAX.CONF***** [general] port=5036 bind=0.0.0.0 bandwidth=low disallow=lpc10 [NuFone] type=peer host=switch-1.nufone.net secret=xxxxxx disallow=all allow=ilbc allow=gsm allow=ulaw disallow=all allow=ulaw allow=alaw allow=gsm ******SIP.CONF***** [general] bindport=5060 bindaddr=0.0.0.0 srvlookup=yes [edward] ;My Xlite softphone type=friend host=dynamic secret=pass-da-word context=from-sip callerid="edward" <100> mailbox=100 disallow=all allow=gsm allow=ulaw allow=alaw allow=ilbc allow=g726 [phone1] ;First analog phone connected to mediatrix type=friend host=dynamic secret=pass-da-word context=from-sip callerid="phone1" <101> mailbox=101 disallow=all allow=gsm allow=ulaw allow=alaw allow=ilbc allow=g726 [phone2] ;Second analog phone connected to mediatrix type=friend host=dynamic secret=pass-da-word context=from-sip callerid="phone2" <102> mailbox=102 disallow=all allow=gsm allow=ulaw allow=alaw allow=ilbc allow=g726 ------------------------------ Message: 12 Date: Fri, 11 Mar 2005 15:57:38 +0530 From: Jagan Mohan <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Load Balancing b/w 2 asterisk servers using SIP load balancer To: Asterisk <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=US-ASCII Hi, I'm trying to do load balancing between 2 asterisk servers using SIP load balancer, provided by http://www.vovida.org I used the following options on lbproxy, but I get the below message continuously. ./lbProxy -name seneca -reqPort 5060 -respPort 5061 -proxy A1 -proxy A2 "No proxies are up - can not send message to anyone" Xlite is not able to register to the asterisk server. Is there anything which needs to be tweaked on Asterisk side to get this working? Please help. Thanks, Jagan ------------------------------ Message: 13 Date: Fri, 11 Mar 2005 11:31:29 +0100 From: "Vledder, Hans" <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] Asterisk and USB ISDN controllers ... To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED] m> Content-Type: text/plain; charset="iso-8859-1" Hi Steve, >Since you don't mention what USB ISDN adapter specifically you are >thinking about, what do you think we will be able to tell you. All I know about the adapter is what I've told you. It's a USB Colognechip based ISDN controller - probably HCF-USB based. It's supported by Linux, but there's no info on access to B and D channels. Regards, Hans -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Thursday, March 10, 2005 6:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and USB ISDN controllers ... On Thu, 2005-03-10 at 18:13 +0100, Vledder, Hans wrote: > Guys, > > I am planning on building a small SIP PBX with a single ISDN line. Currently > I am looking into the specs of a very tiny barebone system that has an > option Colognechip base ISDN controller. The only thing is that the ISDN > module that comes with this barebone hooks up to the motherboard using USB. > My intention is to allow incoming and outgoing calls from SIP to ISDN. Is > this setup in any way supported by *? Since you don't mention what USB ISDN adapter specifically you are thinking about, what do you think we will be able to tell you. The first step would really be to ask if your specific ISDN adapter can be used under linux. After that, can that specific ISDN adapter give access to voice channels. What method is used to get access to the audio and the signaling. It may well be usable if the drivers for it implements the same API as the current ISDN cards in use support. -- Steven Critchfield <[EMAIL PROTECTED]> _______________________________________________ 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 -- The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ------------------------------ Message: 14 Date: Fri, 11 Mar 2005 11:33:33 +0100 From: pbx <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] TE110P experiance To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1; format=flowed [EMAIL PROTECTED] wrote: > > >Hello to all, >I would like to ask some Digium TE110P users if they can share experiance >about this card. I put in service card yesterday but I noticed following >(strange) behaviar: >- if I have to reboot my computer my zaptel driver fail to start and >produce this error: > ZT_SPANCONFIG failed on span 1: No such device or address (6) >- to solve this problem I have to power cycle my computer and in all cases >this brings up card! > >- does anybody have any info about this hardware, example there are two LED >- what is the meaning of these LEDs. I bought this card and got anly card >without any papers (just bill :-( ) >Regards, > >[EMAIL PROTECTED] > >_______________________________________________ >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 > > > This is a well known bug ( can't remeber the number) but the Card Identification on the bus PCI change (for some reason) and the driver is not able to find the card anymore, try to add the line marked with a + in the wcte11cxp.c (in zaptel source), and recompile your driver..... If you like tio verify the bug start your your system from power down do a lspci -v locate "Tiger Jet Network " look the id load zaptel driver do lspci -v ans see the difference Hope this help static struct pci_device_id t1xxp_pci_tbl[] = { { 0xe159, 0x0001, 0x71fe, PCI_ANY_ID, 0, 0, (unsigned long) "Digium Wildcard TE110P T1/E1 Board" }, { 0xe159, 0x0001, 0x79fe, PCI_ANY_ID, 0, 0, (unsigned long) "Digium Wildcard TE110P T1/E1 Board" }, { 0xe159, 0x0001, 0x795e, PCI_ANY_ID, 0, 0, (unsigned long) "Digium Wildcard TE110P T1/E1 Board" }, { 0xe159, 0x0001, 0x79de, PCI_ANY_ID, 0, 0, (unsigned long) "Digium Wildcard TE110P T1/E1 Board" }, + { 0xe159, 0x0001, 0x797e, PCI_ANY_ID, 0, 0, (unsigned long) "Digium Wildcard TE110P T1/E1 Board" }, { 0 } }; ------------------------------ Message: 15 Date: Fri, 11 Mar 2005 11:47:03 +0100 From: "Dennie Verstrepen" <[EMAIL PROTECTED]> Subject: [Asterisk-Users] FW: IAX Settings To: <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" > Hello, > > Has anyone a complete overview of all the settings you can use in the iax.conf file and also where those settings can belong (e.g. in the general section, in a context of type=peer or type=user)? > > Thank you in advance > > Dennie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050311/00 835e2c/attachment-0001.htm ------------------------------ Message: 16 Date: Fri, 11 Mar 2005 10:50:37 +0000 (UTC) From: [EMAIL PROTECTED] (Tony Mountifield) Subject: [Asterisk-Users] Intermittent volume deterioration in conferences To: asterisk-users@lists.digium.com Message-ID: <[EMAIL PROTECTED]> I wonder if anyone can suggest ways to diagnose an infuriating problem being experienced by customers of a company I did a large Asterisk project for. First some background: The system is a conferencing system using a modified MeetMe. There are seven Asterisk boxes (we call them bridges) each with four T1 PRIs into a TE405P. No VoIP is involved. A conference is always local to a single bridge. The conference leader has a control screen and may dial into the bridge, or may instruct the bridge to dial him/her. Once the leader is in the conference, they instruct the bridge to dial each other participant. Each conference is recorded locally in the Asterisk system. The bridges are in Oklahoma and all the leaders and most of the participants are all over Texas. The problem: For the first three or four months of operation everything went very well, but from early February the customer started reporting problems with the volume of audio. Initially the reports seemed to be localized to a particular area of Texas, and to be small in number. Over time, they have increased in frequency and been reported from different areas. Sometimes one participant can't be heard very well by the others, and is also faint on the recording. Other times a participant has trouble hearing the others, but the others are ok on the recording. There does not seem to be any significant distortion, just faint volume. It sounds to me like a phone network issue, but proving that is turning out to be a nightmare. The fact that it is not confined to one bridge but is randomly spread across them would seem to suggest it is not a bridge hardware problem, because it is unlikely to happen in them all. No changes were made to the hardware, Zaptel drivers or Asterisk on the bridges since installation. A day or so ago we disabled echo cancellation on the zap channels, to see if that would make a difference, but it doesn't seem to have. It still wouldn't explain why the problem did not previously exist, and started happening spontaneously. Sometimes if it's really difficult for people to hear, the leader closes the conference and reverts to their older conferencing system (that our system replaced), and reports that the volume is then fine. I don't know where the older system is located, but I believe it is more local. This is obviously a worrying scenario. If anyone can suggest any ideas of ways to tackle the problem, and to determine whether it really is the Asterisk bridges or the phone systems, I would be very, very grateful, as it is turning into a nightmare! Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ------------------------------ Message: 17 Date: Fri, 11 Mar 2005 12:02:21 +0100 From: Giovanni Miano <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Asterisk + Call hangup To: Asterisk-Users@lists.digium.com Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=US-ASCII Scenario PSTN <-> ZAP CHANNEL <-> ASTERISK <-> SIP When i recive call i fwd it to SIP Phone -> SIP PHONE ringing If From External Line PSTN hungup call SIP Phone Ringing too, why ? ------------------------------ _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest, Vol 8, Issue 88 ********************************************* _______________________________________________ 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
