Re: [Asterisk-Users] unable to configure my Grandstream phone
i hv also added the alaw > [general] > port = 5060 > bindaddr = 0.0.0.0 > context = bogon-calls > ;context = default > disallow=all > allow=g729 > allow=gsm > allow=ulaw allow=alaw now i am able to call from my MSN -> * ->GS but the other way is not working. i am getting lot of noise when i try to place any call from my GS. i am no longer getting codec not compatible error message anymore. i am still unable to place any calls using my GS (to my internal MSN extensions or to external PSTN). thanks for ur help, -B > - Original Message - From: "Balaji NJL" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Cc: <[EMAIL PROTECTED]> Sent: Sunday, December 14, 2003 3:40 PM Subject: Re: [Asterisk-Users] unable to configure my Grandstream phone > Hi Paul, > > thanks for the quick response. i tried the following > configuration / > combination still no luck > > [general] > port = 5060 > bindaddr = 0.0.0.0 > context = bogon-calls > ;context = default > disallow=all > allow=g729 > allow=gsm > allow=ulaw > > when i tried g711 i am getting an error in * that > codec not found. > > When i specify only g729 my MSN doesnt work. > > thanks, > -B > > - Original Message - > From: "Paul Liew" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Sunday, December 14, 2003 2:14 PM > Subject: Re: [Asterisk-Users] unable to configure my > Grandstream phone > > > > > > - Original Message - > > From: Balaji NJL > > To: [EMAIL PROTECTED] > > Sent: Monday, December 15, 2003 8:47 AM > > Subject: [Asterisk-Users] unable to configure my > Grandstream phone > > > > > > > > > Attempting native bridge of SIP/2003-b895 and > SIP/2000-53e2 > > > WARNING[5126]: File chan_sip.c, Line 1954 > (process_sdp): No compatible > > codecs! > > > -- Got SIP response 481 "Call Leg/Transaction > Does Not Exist" back > > from 192.168.0.58 > > > > > > and then the call drops. When i am making a call > using Grandstream ph, > it > > rings the other side when they pick up the phone the > call then drops. then > i > > get the > > > above error message. > > > > > > the follwoing us sip and Grandstream conf > > > > > > > > > [general] > > > port = 5060 > > > bindaddr = 0.0.0.0 > > > context = bogon-calls > > > ;context = default > > > disallow=all > > > allow=gsm > > > > Balaji, > > > > Grandstreams do not support GSM. Options available > can be seen on the GS > > config page. Unless you purchase G729s (for low > bandwidth), your only > choice > > is G711. > > > > Paul > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > __ > Do you Yahoo!? > New Yahoo! Photos - easier uploading and sharing. > http://photos.yahoo.com/ > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] unable to configure my Grandstream phone
Hi Paul, thanks for the quick response. i tried the following configuration / combination still no luck [general] port = 5060 bindaddr = 0.0.0.0 context = bogon-calls ;context = default disallow=all allow=g729 allow=gsm allow=ulaw when i tried g711 i am getting an error in * that codec not found. When i specify only g729 my MSN doesnt work. thanks, -B - Original Message - From: "Paul Liew" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Sunday, December 14, 2003 2:14 PM Subject: Re: [Asterisk-Users] unable to configure my Grandstream phone > > - Original Message - > From: Balaji NJL > To: [EMAIL PROTECTED] > Sent: Monday, December 15, 2003 8:47 AM > Subject: [Asterisk-Users] unable to configure my Grandstream phone > > > > > Attempting native bridge of SIP/2003-b895 and SIP/2000-53e2 > > WARNING[5126]: File chan_sip.c, Line 1954 (process_sdp): No compatible > codecs! > > -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back > from 192.168.0.58 > > > > and then the call drops. When i am making a call using Grandstream ph, it > rings the other side when they pick up the phone the call then drops. then i > get the > > above error message. > > > > the follwoing us sip and Grandstream conf > > > > > > [general] > > port = 5060 > > bindaddr = 0.0.0.0 > > context = bogon-calls > > ;context = default > > disallow=all > > allow=gsm > > Balaji, > > Grandstreams do not support GSM. Options available can be seen on the GS > config page. Unless you purchase G729s (for low bandwidth), your only choice > is G711. > > Paul > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] unable to configure my Grandstream phone
- Original Message - From: Balaji NJL To: [EMAIL PROTECTED] Sent: Monday, December 15, 2003 8:47 AM Subject: [Asterisk-Users] unable to configure my Grandstream phone > Attempting native bridge of SIP/2003-b895 and SIP/2000-53e2 > WARNING[5126]: File chan_sip.c, Line 1954 (process_sdp): No compatible codecs! > -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 192.168.0.58 > > and then the call drops. When i am making a call using Grandstream ph, it rings the other side when they pick up the phone the call then drops. then i get the > above error message. > > the follwoing us sip and Grandstream conf > > > [general] > port = 5060 > bindaddr = 0.0.0.0 > context = bogon-calls > ;context = default > disallow=all > allow=gsm Balaji, Grandstreams do not support GSM. Options available can be seen on the GS config page. Unless you purchase G729s (for low bandwidth), your only choice is G711. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] unable to configure my Grandstream phone
Hi All, i received my X100P and Grandstream phone last week. i started configuring my * and with the help of ur mailing lists i was able to configure it. (when ever i got struck i searched this list and found my answer. thanks a lot and this list is awesome). i still hv a small problem and hope someone could help me out. This is my setup. RH 7.2 serving as my * server. i hv got couple of my laptops and desktops running MSN 4.7. I hv installed and configured X100P and Grandstream phone the following configurations are working MSN Msgr -> * -> MSN Msgr MSN Msgr -> * -> X100P - PSTN PSTN -> * -> MSN Msgr PSTN -> * -> Grandstream (pl note) the following are *not* working MSN Msgr -> * -> Grandstream Grandstream -> * -> MSN Msgr GrandStream -> * -> PSTN PSTN -> * -> Grandstream the error i am getting in this case is - Attempting native bridge of SIP/2003-b895 and SIP/2000-53e2WARNING[5126]: File chan_sip.c, Line 1954 (process_sdp): No compatible codecs! -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 192.168.0.58 and then the call drops. When i am making a call using Grandstream ph, it rings the other side when they pick up the phone the call then drops. then i get the above error message. the follwoing us sip and Grandstream conf [general]port = 5060bindaddr = 0.0.0.0context = bogon-calls;context = defaultdisallow=allallow=gsm [2000]; Grandstream phone type=friendusername=2000secret=qweqwehost=dynamiccontext=from-sipmailbox=2000dtmfmode=inband [2002] type=friendhost=dynamicinsecure=yesdtmfmode=inband;dtmfmode=rfc2833context=from-sipmailbox=2002;auth=plaintext [2003] type=friendhost=dynamicinsecure=yesdtmfmode=inbandcontext=from-sipmailbox=2003 ;[2000] ;type=friend;username=2000;secret=qweqwe;auth=md5;host=dynamic;context=from-sip;dtmfmode=inband;mailbox=2000 [2001] type=friendusername=2001secret=asdasdauth=md5host=dynamiccontext=from-sipdtmfmode=inbandmailbox=2001 Grandstream configuration details >SIP Server: 192.168.0.4 (my * box)>SIP Userid: 2000 (userid is same as extension>Authenticate ID: 2000>Authenticate password: qweqwe >Send DTMF: Via SIP info (in order for the dtmf to be recognized by>voicemail)>> Program--1.0.4.17 Bootloader--1.0.0.11 HTML--1.0.0.19 any idea why my Grandstream drops the calls. thanks a lot and appreciate ur help. -B Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing
