RE: [Asterisk-Users] Handytone 286 - calling out
Billy Huddleston wrote: > I've not had ANY problems using info OR rfc2833.. I did have problems > using inband. Try switching to it and see how it works.. I NEVER > had a problem with double digits, and, I believe that the reference > to GS phones having that problem with * was retracted. > > Thanks, Billy > > - Original Message - > From: "Senad Jordanovic" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Wednesday, November 26, 2003 4:14 AM > Subject: RE: [Asterisk-Users] Handytone 286 - calling out > > >> Billy Huddleston wrote: >>> change dtmf to info on both * and in the handytone. >>> >>> - Original Message - >>> From: "Senad Jordanovic" <[EMAIL PROTECTED]> >>> To: <[EMAIL PROTECTED]> >>> Sent: Tuesday, November 25, 2003 8:01 PM >>> Subject: [Asterisk-Users] Handytone 286 - calling out >>> >>> >>>> Hi, >>>> >>>> Just received recently released Grandstream handytone 286 ATA for >>>> testing. >>>> >>>> I can call ATA from any other extensions and conversations seems to >>>> be of quite good quality. However placing calls from ATA is not >>>> possible at all to any extensions. After dialing there no >>>> indications of any kind from ATA at all. It just "hangs in there". >>>> >>>> ATA is behind NAT, registers to an * with public IP with no >>>> problems and it uses 1.0.4.17 firmware. Web config screen has >>>> detected "firewall/NAT type is open Internet" as network firewall. >>>> >>>> Here is my sip.conf: >>>> [2202] >>>> callerid="HandyTone" <2202> >>>> username=2202 >>>> context=intern >>>> qualify=500 >>>> type=friend >>>> secret=XX >>>> host=dynamic >>>> dtmfmode=inband >>>> canreinvite=no >>>> reinvite=no >>>> disallow=all >>>> allow=ulaw >>>> allow=alaw >>>> >>>> Any suggestions/pointers will be appreciated. >>>> >>>> Ta >>>> SJ >>>> >>>> ___ >>>> Asterisk-Users mailing list [EMAIL PROTECTED] >>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>> >>> ___ >>> Asterisk-Users mailing list >>> [EMAIL PROTECTED] >>> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> My understanding from this months GS related posts is that "info" is >> not sending the digits properly. Is that the case with you? >> >> ___ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users Yes, same here... Thanks for sharing... SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Handytone 286 - calling out
I've not had ANY problems using info OR rfc2833.. I did have problems using inband. Try switching to it and see how it works.. I NEVER had a problem with double digits, and, I believe that the reference to GS phones having that problem with * was retracted. Thanks, Billy - Original Message - From: "Senad Jordanovic" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, November 26, 2003 4:14 AM Subject: RE: [Asterisk-Users] Handytone 286 - calling out > Billy Huddleston wrote: > > change dtmf to info on both * and in the handytone. > > > > - Original Message - > > From: "Senad Jordanovic" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Sent: Tuesday, November 25, 2003 8:01 PM > > Subject: [Asterisk-Users] Handytone 286 - calling out > > > > > >> Hi, > >> > >> Just received recently released Grandstream handytone 286 ATA for > >> testing. > >> > >> I can call ATA from any other extensions and conversations seems to > >> be of quite good quality. However placing calls from ATA is not > >> possible at all to any extensions. After dialing there no > >> indications of any kind from ATA at all. It just "hangs in there". > >> > >> ATA is behind NAT, registers to an * with public IP with no problems > >> and it uses 1.0.4.17 firmware. Web config screen has detected > >> "firewall/NAT type is open Internet" as network firewall. > >> > >> Here is my sip.conf: > >> [2202] > >> callerid="HandyTone" <2202> > >> username=2202 > >> context=intern > >> qualify=500 > >> type=friend > >> secret=XX > >> host=dynamic > >> dtmfmode=inband > >> canreinvite=no > >> reinvite=no > >> disallow=all > >> allow=ulaw > >> allow=alaw > >> > >> Any suggestions/pointers will be appreciated. > >> > >> Ta > >> SJ > >> > >> ___ > >> Asterisk-Users mailing list > >> [EMAIL PROTECTED] > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > My understanding from this months GS related posts is that "info" is not > sending the digits properly. > Is that the case with you? > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Handytone 286 - calling out
Billy Huddleston wrote: > change dtmf to info on both * and in the handytone. > > - Original Message - > From: "Senad Jordanovic" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Tuesday, November 25, 2003 8:01 PM > Subject: [Asterisk-Users] Handytone 286 - calling out > > >> Hi, >> >> Just received recently released Grandstream handytone 286 ATA for >> testing. >> >> I can call ATA from any other extensions and conversations seems to >> be of quite good quality. However placing calls from ATA is not >> possible at all to any extensions. After dialing there no >> indications of any kind from ATA at all. It just "hangs in there". >> >> ATA is behind NAT, registers to an * with public IP with no problems >> and it uses 1.0.4.17 firmware. Web config screen has detected >> "firewall/NAT type is open Internet" as network firewall. >> >> Here is my sip.conf: >> [2202] >> callerid="HandyTone" <2202> >> username=2202 >> context=intern >> qualify=500 >> type=friend >> secret=XX >> host=dynamic >> dtmfmode=inband >> canreinvite=no >> reinvite=no >> disallow=all >> allow=ulaw >> allow=alaw >> >> Any suggestions/pointers will be appreciated. >> >> Ta >> SJ >> >> ___ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users My understanding from this months GS related posts is that "info" is not sending the digits properly. Is that the case with you? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Handytone 286 - calling out
Andrew Gillham wrote: > Senad Jordanovic wrote: > >> Hi, >> >> Just received recently released Grandstream handytone 286 ATA for >> testing. >> >> I can call ATA from any other extensions and conversations seems to >> be of quite good quality. However placing calls from ATA is not >> possible at all to any extensions. After dialing there no >> indications of any kind from ATA at all. It just "hangs in there". >> >> ATA is behind NAT, registers to an * with public IP with no problems >> and it uses 1.0.4.17 firmware. Web config screen has detected >> "firewall/NAT type is open Internet" as network firewall. >> >> >> > Are you able to use tcpdump on the asterisk box to capture traffic > from the ATA? Or Ethereal if you have X installed on the Linux box. > > It would be interesting to see if the ATA sends anything after you > dial the '1234#' sequence. > > On my Grandstream 101 phone I have not had any trouble placing calls. > I don't have the 'Outbound Proxy' field configured, and I re-ordered > the codec preferences as well. Other than that it is pretty much > stock with SIP server / user and authentication configured. > > -Andrew > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users I have tested several GS 101s as well, and they worked. I will try to capture traffic, then we will see from there. Ta ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Handytone 286 - calling out
change dtmf to info on both * and in the handytone. - Original Message - From: "Senad Jordanovic" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, November 25, 2003 8:01 PM Subject: [Asterisk-Users] Handytone 286 - calling out > Hi, > > Just received recently released Grandstream handytone 286 ATA for > testing. > > I can call ATA from any other extensions and conversations seems to be > of quite good quality. However placing calls from ATA is not possible at > all to any extensions. > After dialing there no indications of any kind from ATA at all. It just > "hangs in there". > > ATA is behind NAT, registers to an * with public IP with no problems and > it uses 1.0.4.17 firmware. Web config screen has detected "firewall/NAT > type is open Internet" as network firewall. > > Here is my sip.conf: > [2202] > callerid="HandyTone" <2202> > username=2202 > context=intern > qualify=500 > type=friend > secret=XX > host=dynamic > dtmfmode=inband > canreinvite=no > reinvite=no > disallow=all > allow=ulaw > allow=alaw > > Any suggestions/pointers will be appreciated. > > Ta > SJ > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Handytone 286 - calling out
Senad Jordanovic wrote: Hi, Just received recently released Grandstream handytone 286 ATA for testing. I can call ATA from any other extensions and conversations seems to be of quite good quality. However placing calls from ATA is not possible at all to any extensions. After dialing there no indications of any kind from ATA at all. It just "hangs in there". ATA is behind NAT, registers to an * with public IP with no problems and it uses 1.0.4.17 firmware. Web config screen has detected "firewall/NAT type is open Internet" as network firewall. Are you able to use tcpdump on the asterisk box to capture traffic from the ATA? Or Ethereal if you have X installed on the Linux box. It would be interesting to see if the ATA sends anything after you dial the '1234#' sequence. On my Grandstream 101 phone I have not had any trouble placing calls. I don't have the 'Outbound Proxy' field configured, and I re-ordered the codec preferences as well. Other than that it is pretty much stock with SIP server / user and authentication configured. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Handytone 286 - calling out
Hi, Just received recently released Grandstream handytone 286 ATA for testing. I can call ATA from any other extensions and conversations seems to be of quite good quality. However placing calls from ATA is not possible at all to any extensions. After dialing there no indications of any kind from ATA at all. It just "hangs in there". ATA is behind NAT, registers to an * with public IP with no problems and it uses 1.0.4.17 firmware. Web config screen has detected "firewall/NAT type is open Internet" as network firewall. Here is my sip.conf: [2202] callerid="HandyTone" <2202> username=2202 context=intern qualify=500 type=friend secret=XX host=dynamic dtmfmode=inband canreinvite=no reinvite=no disallow=all allow=ulaw allow=alaw Any suggestions/pointers will be appreciated. Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
