Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
On Thu, 20 Nov 2003, marrandy wrote: On Monday 17 November 2003 10:31 pm, Brian West wrote: Show us your sip.conf entries.. and i'm sure I can point out the error. bkw Well, I've tried over 20 different settings, from examples in the archives etc. This is the last one I tried, for what it's worth. I had strange registration problems with my Grandstream SIP phones which vanished when I changed from specifying the host in sip.conf to host=dynamic defaultip=192.168.254.160 Not sure if that is at all helpful, but it was rather peculiar. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
*CLI NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.70' I've seen this in only two cases: 1) when the SIP user ID, the Authenticate ID (on the GS) and the extension name in sip.conf are not all the same, in your case, 206. The user name in sip.conf doesn't matter, and in fact doesent seem to do anything at all with the GS. 2) You are using a secret and the authenticate Password does not match the secret. This does not appear to be relevant to your situation. You might also check to be sure that the registration option in the GS is turned off, since you are hard coding the IP address. There is a bug in the current GS firmware (supposed to be fixed soon) that sometimes messes up the registration renewal. -- Username not entered -- Executing Hangup(SIP/206-7ecb, ) in new stack You can't use INFO with the GS. * and GS interpret the INFO standard differently. As a result, the GS does multiple digit transmission. Use either inband or the RFC2833 option. INFO will not work no matter what you do. Also, if you are hard coding your phone's IP addresses, then the permit option has no meaning. [206] username=206 context=extensions qualify=yes incominglimit=1 type=friend insecure=yes host=192.168.1.70 permit=192.168.1.0/255.255.255.0 dtmfmode=info canreinvite=no reinvite=no callgroup=1 pickupgroup=1 disallow=all allow=alaw allow=ulaw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
You can't use INFO with the GS. * and GS interpret the INFO standard differently. As a result, the GS does multiple digit transmission. Use either inband or the RFC2833 option. INFO will not work no matter what you do. I am using the following settings Software Version:Program--1.0.4.20Bootloader--1.0.0.12 HTML--1.0.0.19 Send DTMF: via SIP INFO DTMF Payload Type: 101 and I can dial dtmf fine interact with * Background menus other PBX IVR systems What test are you doing to show INFO is not supported on GS ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
On Fri, 2003-11-21 at 17:57, TC wrote: I am using the following settings Software Version:Program--1.0.4.20Bootloader--1.0.0.12 HTML--1.0.0.19 Where is this firmware from? The GS site is still at Program--1.0.3.81Bootloader--1.0.0.7HTML--1.0.0.18 or is Sipphone ahead of GS? -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
TC wrote: You can't use INFO with the GS. * and GS interpret the INFO standard differently. As a result, the GS does multiple digit transmission. Use either inband or the RFC2833 option. INFO will not work no matter what you do. I am using the following settings Software Version:Program--1.0.4.20Bootloader--1.0.0.12 HTML--1.0.0.19 Send DTMF: via SIP INFO DTMF Payload Type: 101 and I can dial dtmf fine interact with * Background menus other PBX IVR systems What test are you doing to show INFO is not supported on GS It's not that INFO is not supported, it's just that it did not work without transmitting multiple copies of digits to *. I have been told (by GS) that this was a problem with *, and it may be that it has been fixed in more recent CVS releases (mine is about 2 months old right now). On the otherhand, unless I use inband, the # key does not work on some (not all) IVR systems that I dial into. Rather than wasting time trying to figure out why, I just use inband. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
Good day, I have a new Grandstream and am having trouble connecting to * my software version is the same as below... I can get it to connect, but am getting RTP Read error: Resource temporarily unavailable errors whenever I dial... Tom - Original Message - From: Dave Cotton [EMAIL PROTECTED] To: Asterisk List [EMAIL PROTECTED] Sent: Friday, November 21, 2003 10:15 AM Subject: Re: [Asterisk-Users] Struggling with grandstream sip to asterisk On Fri, 2003-11-21 at 17:57, TC wrote: I am using the following settings Software Version:Program--1.0.4.20Bootloader--1.0.0.12 HTML--1.0.0.19 Where is this firmware from? The GS site is still at Program--1.0.3.81Bootloader--1.0.0.7HTML--1.0.0.18 or is Sipphone ahead of GS? -- Dave Cotton [EMAIL PROTECTED] ___ 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] Struggling with grandstream sip to asterisk
On Friday 21 November 2003 06:17 am, Michael T Farnworth wrote: On Thu, 20 Nov 2003, marrandy wrote: host=dynamic defaultip=192.168.254.160 O.K. - that worked, plus I removed the permit and went back to inband (or grandstream calls it, in-audio. So I can make and receive calls. My next observation, is many of the buttons on the phone don't work e.g. Called - After make calls, I expected that pressing this button would show something. Nothing is shown. Callers - After the grandstream receives calls, I expected that pressing this button would show something. Nothing is shown Hold - Pressing hold during a call does nothing. Transfer - pressing transfer does nothing. Conference - does nothing flash -does nothing redial - does nothing Message button - does nothing, even when message are waiting (stutter dialtone. Any clues on getting these features working ? Regards...Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
Software Version:Program--1.0.4.20Bootloader--1.0.0.12 HTML--1.0.0.19 Send DTMF: via SIP INFO DTMF Payload Type: 101 In my experience, anything higher then 1.0.4.17 will NOT work properly with asterisk. You can get the phone to sip register but it will not stay that way. The tftp site you are using is BETA software if I'm not mistaken. Correct, never the less I do not see the same issues other are reporting with SIP not staying registered or the SIP Info issue with the DTMF How can I verify this registration issue, its werid that it just seems to work in my install ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
On Fri, 2003-11-21 at 20:14, marrandy wrote: My next observation, is many of the buttons on the phone don't work e.g. Called - After make calls, I expected that pressing this button would show something. Nothing is shown. Pickup and press called the last 10 numbers called are shown Callers - After the grandstream receives calls, I expected that pressing this button would show something. Nothing is shown As above redial - does nothing redials last number or saves the delay after dialling Message button - does nothing, even when message are waiting (stutter dialtone. Program with your voice mail extension number ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
On Friday 21 November 2003 03:22 pm, Dave Cotton wrote: On Fri, 2003-11-21 at 20:14, marrandy wrote: My next observation, is many of the buttons on the phone don't work e.g. Called - After make calls, I expected that pressing this button would show something. Nothing is shown. Pickup and press called the last 10 numbers called are shown That's what I am doing - Nothing is shown Callers - After the grandstream receives calls, I expected that pressing this button would show something. Nothing is shown As above As above, I have tried it both ways. Nothing is shown. redial - does nothing redials last number or saves the delay after dialling Nothing happens. Message button - does nothing, even when message are waiting (stutter dialtone. Program with your voice mail extension number Done...it doesn't work. Perhap's my phone is just defective on the buttons. Regards...Martin -- The secret of success is sincerity. Once you can fake that, you've got it made. -- Jean Giraudoux ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
Correct, never the less I do not see the same issues other are reporting with SIP not staying registered I only saw this after I had installed 20 phones. When I had only one phone installed during the test and setup phase, it never happened. Either it is rather infrequent, or you have the re-registration interval set really low, or it has to do with more than one phone trying to register at once. One never knows! or the SIP Info issue with the DTMF Alas, maybe I was just too damn impatient to struggle with getting anything but in-band to work. How can I verify this registration issue, its werid that it just seems to work in my install ... Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
On Monday 17 November 2003 10:31 pm, Brian West wrote: Show us your sip.conf entries.. and i'm sure I can point out the error. bkw Hello Brian. Well, I've tried over 20 different settings, from examples in the archives etc. This is the last one I tried, for what it's worth. Grandstream Program--1.0.3.81Bootloader--1.0.0.7HTML--1.0.0.18 Grandstream is 192.1681.170 asterisk is 192.168.1.1. What happens presently, is I call call from/to an analog phone on zap/2-1 (fxo, tdm400). If I try voicemail, I get the comdian mail prompt. But when I enter an extension, it times out. Debug says:- *CLI sip show peers Name/usernameHost Mask Port Status 206/206 192.168.1.70 255.255.255.255 5060 OK (13 ms) *CLI NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.70' NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.70' NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.70' NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.70' NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.70' -- Executing VoiceMailMain(SIP/206-7ecb, ) in new stack == Parsing '/etc/asterisk/voicemail.conf': Found -- Playing 'vm-login' NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.70' NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.70' NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.70' NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.70' NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.70' -- Username not entered -- Executing Hangup(SIP/206-7ecb, ) in new stack == Spawn extension (extensions, 15, 2) exited non-zero on 'SIP/206-7ecb' NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.70' NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.70' contd... So basically, I am seeing Two different errors. 1) registration errors with grandstream 2) Voicemail doesn't read the extension tones from the grandstream Voicemail works fine from the analog phone. SIP.conf [general] port=5060 bindaddr=0.0.0.0 context=incoming ; ; ; [206] username=206 context=extensions qualify=yes incominglimit=1 type=friend insecure=yes host=192.168.1.70 permit=192.168.1.0/255.255.255.0 dtmfmode=info canreinvite=no reinvite=no callgroup=1 pickupgroup=1 disallow=all allow=alaw allow=ulaw Regards...Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
On Monday 17 November 2003 10:31 pm, Brian West wrote: Show us your sip.conf entries.. and i'm sure I can point out the error. bkw Follow-up. I think there are a number of issues, possibly conficting. Let me ask some specific questions relating to grandstream to asterisk operation. grandstream = 192.168.1.70 asterisk = 192.168.1.1 1) What DTMF should I be using on the grandstream ? in-audio(their term, I assume that means in-band ?), RTP (RFC2833) or SIP INFO 2) What is the minimum sip.conf file that should achieve full inter-operability with errors and problems ? Regards...Martin PS. There seems to be a 2-hour lag in mail to this list at the moment, based on the last few of mails I received. -- Phasers locked on target, Captain. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Struggling with grandstream sip to asterisk
Hello. I had grandstream working fine to FWD through my firewall. Now I want it to talk to the asterisk server. Did lots of reading, attempts but I keep getting registration errors even though I can call to/from the sip phone from an analog phone on a tdm400 card. Basically. grandstream = 192.168.1.70 asterisk = 192.168.1.1 The error I see is ;- -- Executing Dial(Zap/2-1, SIP/206|20) in new stack -- Called 206 -- SIP/206-582e is ringing NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.70' Can anyone give me a hint/clue about what is going on. A good sip.conf for grandstream with a static address would help. Regards...Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Struggling with grandstream sip to asterisk
I just got my GrandStream working. The SIP.CONF file probably isn't your problem except for the 'secret' entry. Your phone doesn't have the proper auth/password combination to successfully register with the Asterisk Server. Use the web browser for your GrandStream phone to set the 'Authenticate ID' and 'Authenticate Password' to match what you've configured in your sip.conf file. My sip.conf looks like this: [7007] type=friend secret=blah host=dynamic mailbox=7002 canreinvite=no The fact that you are using a Static IP Address doesn't matter to the *, so don't worry about it. Once configured, I've found my GrandStream to provide decent voice quality. It doesn't have all of the features of a more expensive phone, but then that isn't what you paid for. Joe Dennick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of marrandy Sent: Monday, November 17, 2003 8:56 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Struggling with grandstream sip to asterisk Hello. I had grandstream working fine to FWD through my firewall. Now I want it to talk to the asterisk server. Did lots of reading, attempts but I keep getting registration errors even though I can call to/from the sip phone from an analog phone on a tdm400 card. Basically. grandstream = 192.168.1.70 asterisk = 192.168.1.1 The error I see is ;- -- Executing Dial(Zap/2-1, SIP/206|20) in new stack -- Called 206 -- SIP/206-582e is ringing NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.70' Can anyone give me a hint/clue about what is going on. A good sip.conf for grandstream with a static address would help. Regards...Martin ___ 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] Struggling with grandstream sip to asterisk
Basically. grandstream = 192.168.1.70 asterisk = 192.168.1.1 The error I see is ;- -- Executing Dial(Zap/2-1, SIP/206|20) in new stack -- Called 206 -- SIP/206-582e is ringing NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.70' Can anyone give me a hint/clue about what is going on. A good sip.conf for grandstream with a static address would help. ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls ; [205] ; Conference 2, Grandstream Phone callerid=Converence 2 205 username=205 context=intern qualify=yes incominglimit=1 type=friend insecure=yes host=192.168.1.70 permit=192.168.0.0/255.255.255.0 dtmfmode=info canreinvite=no reinvite=no callgroup=1 pickupgroup=1 disallow=all allow=alaw allow=ulaw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
Show us your sip.conf entries.. and i'm sure I can point out the error. bkw On Mon, 17 Nov 2003, marrandy wrote: Hello. I had grandstream working fine to FWD through my firewall. Now I want it to talk to the asterisk server. Did lots of reading, attempts but I keep getting registration errors even though I can call to/from the sip phone from an analog phone on a tdm400 card. Basically. grandstream = 192.168.1.70 asterisk = 192.168.1.1 The error I see is ;- -- Executing Dial(Zap/2-1, SIP/206|20) in new stack -- Called 206 -- SIP/206-582e is ringing NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.70' Can anyone give me a hint/clue about what is going on. A good sip.conf for grandstream with a static address would help. Regards...Martin ___ 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] Struggling with grandstream sip to asterisk
quote who=Walker Haddock ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls ; [205] ; Conference 2, Grandstream Phone callerid=Converence 2 205 username=205 context=intern qualify=yes incominglimit=1 type=friend insecure=yes host=192.168.1.70 permit=192.168.0.0/255.255.255.0 ^ wrong subnet. dtmfmode=info canreinvite=no reinvite=no callgroup=1 pickupgroup=1 disallow=all allow=alaw allow=ulaw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
