Re: [asterisk-users] Incoming SIP call, outgoing SIP registration. PJSIP.
23.04.2019 0:27, Joshua C. Colp wrote: On Mon, Apr 22, 2019, at 2:10 PM, Pavel wrote: Tried already. "line" is good, but not perfect. Every time I restart asterisk, it will generate new random string for ";line=". So, every time I restart asterisk, registrar (Server1) will save one more contact in it's database. Some will remove obsolete contacts, but some will not. For example, FreePBX will not remove obsolete contacts, if max_contacts specified (FreePBX will set rewrite_contact=no in this case). So, after a number of Asterisk restarts, FreePBX will reject new registrations, as max_contacts is reached. It should specify remove_existing to remove old ones to make room for the new ones. That would be a FreePBX thing, though. FreePBX is an example, where it can be a critical problem. 3cx will work, but if you will restart asterisk 10 times - you will see 10 times more contacts in 3cx. When you will make call from 3cx - it will make 10 calls (10 contacts), untill they will obsolete... Unfortunately, "line" does not save random between restarts. It's also unable to specify "random" value in pjsip.conf. I'm thinking to patch res_pjsip_outbound_registration to add this feature. Am I wrong and there is another way ? I don't see any reason why this couldn't be an option. For flexibility. Not to register new fake contacts in peer PBX. It's also a security hole, as anybody can generate INVITE with ";line=random" from any IP address ! You can use an ACL to limit the endpoint to certain source IP addresses. 5+ ! Thank you, ACL is a good idea ! res_pjsip_outbound_registration will only match "line", but will not take care about source IP, ... Is there any more clear way to identify incoming INVITE/OPTIONS packets ? Not very familliar with SIP, not sure, how should it be done. There is no real defined mechanism within SIP to do this. Phones employ different mechanisms to differentiate. Some may use a similar mechanism to the line option. Some run multiple SIP transports on different ports for each account so they can differentiate based on where it came in on. Some look at the request URI coming in. Some just don't care. Sniffered some time ago how it's done in phonerlite, jitsi, linksys, ... Some use different port, some use ";rinstance=", the same like ";line=" in asterisk. Was not sure it's a right way to go. I will probably extend "line" a bit to specify it's value in pjsip.conf . It will be less than 10 lines of code. Thank you very much ! Your help will simplify my life a lot :-) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP call, outgoing SIP registration. PJSIP.
On Mon, Apr 22, 2019, at 2:10 PM, Pavel wrote: > Tried already. > > "line" is good, but not perfect. > > Every time I restart asterisk, it will generate new random string for > ";line=". > > So, every time I restart asterisk, registrar (Server1) will save one > more contact in it's database. > > Some will remove obsolete contacts, but some will not. > > For example, FreePBX will not remove obsolete contacts, if max_contacts > specified (FreePBX will set rewrite_contact=no in this case). > > So, after a number of Asterisk restarts, FreePBX will reject new > registrations, as max_contacts is reached. It should specify remove_existing to remove old ones to make room for the new ones. That would be a FreePBX thing, though. > Unfortunately, "line" does not save random between restarts. > > It's also unable to specify "random" value in pjsip.conf. > > > I'm thinking to patch res_pjsip_outbound_registration to add this feature. > > Am I wrong and there is another way ? I don't see any reason why this couldn't be an option. > > It's also a security hole, as anybody can generate INVITE with > ";line=random" from any IP address ! You can use an ACL to limit the endpoint to certain source IP addresses. > > res_pjsip_outbound_registration will only match "line", but will not > take care about source IP, ... > > > > Is there any more clear way to identify incoming INVITE/OPTIONS packets ? > > Not very familliar with SIP, not sure, how should it be done. There is no real defined mechanism within SIP to do this. Phones employ different mechanisms to differentiate. Some may use a similar mechanism to the line option. Some run multiple SIP transports on different ports for each account so they can differentiate based on where it came in on. Some look at the request URI coming in. Some just don't care. -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP call, outgoing SIP registration. PJSIP.
Hi, Thank for your answer. 22.04.2019 23:47, Joshua C. Colp пишет: On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote: Hi, Got problems with incoming SIP calls. Scenario: Server1: 3cx or any other server Server2: Asterisk 16.2.1 . PJPROJECT 2.8 Server2 registers on Server1 with SIP ID 1121. Registration is OK. Server2 outgoing calls are OK. INVITE, unauthorized, INVITE with password, OK, RINGING,... Troubles with incoming calls / incoming INVITE's . I can not identify endpoint by IP, I have multiple registrations on the same Server1. As far as I understood, res_pjsip_endpoint_identifier_user match endpoint by "From" header, so it will not match also. match_headers also seems useless (not able to match "INVITE" string, just headers like "TO:"). Is there any way to match incoming INVITE calls ? (also OPTIONS, NOTIFY, ... packets) It should be a typical scenario, but it does not work... Is there any way to make it working ? Outbound registration provides the line option[1] which can be used to differentiate traffic in regards to different outbound registrations. It requires the remote server to adhere to the SIP RFC and report back some data we give in our Contact, so you have to test it and see if it works. [1] https://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/ Tried already. "line" is good, but not perfect. Every time I restart asterisk, it will generate new random string for ";line=". So, every time I restart asterisk, registrar (Server1) will save one more contact in it's database. Some will remove obsolete contacts, but some will not. For example, FreePBX will not remove obsolete contacts, if max_contacts specified (FreePBX will set rewrite_contact=no in this case). So, after a number of Asterisk restarts, FreePBX will reject new registrations, as max_contacts is reached. Unfortunately, "line" does not save random between restarts. It's also unable to specify "random" value in pjsip.conf. I'm thinking to patch res_pjsip_outbound_registration to add this feature. Am I wrong and there is another way ? It's also a security hole, as anybody can generate INVITE with ";line=random" from any IP address ! res_pjsip_outbound_registration will only match "line", but will not take care about source IP, ... Is there any more clear way to identify incoming INVITE/OPTIONS packets ? Not very familliar with SIP, not sure, how should it be done. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP call, outgoing SIP registration. PJSIP.
On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote: > Hi, > > Got problems with incoming SIP calls. > > Scenario: > > Server1: 3cx or any other server > > Server2: Asterisk 16.2.1 . PJPROJECT 2.8 > > Server2 registers on Server1 with SIP ID 1121. > > Registration is OK. > > Server2 outgoing calls are OK. > > INVITE, unauthorized, INVITE with password, OK, RINGING,... > > Troubles with incoming calls / incoming INVITE's . > > I can not identify endpoint by IP, I have multiple registrations on the > same Server1. > > As far as I understood, res_pjsip_endpoint_identifier_user match > endpoint by "From" header, so it will not match also. > > match_headers also seems useless (not able to match "INVITE" string, > just headers like "TO:"). > > Is there any way to match incoming INVITE calls ? (also OPTIONS, NOTIFY, > ... packets) > > It should be a typical scenario, but it does not work... > > Is there any way to make it working ? Outbound registration provides the line option[1] which can be used to differentiate traffic in regards to different outbound registrations. It requires the remote server to adhere to the SIP RFC and report back some data we give in our Contact, so you have to test it and see if it works. [1] https://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/ -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming SIP call, outgoing SIP registration. PJSIP.
Hi, Got problems with incoming SIP calls. Scenario: Server1: 3cx or any other server Server2: Asterisk 16.2.1 . PJPROJECT 2.8 Server2 registers on Server1 with SIP ID 1121. Registration is OK. Server2 outgoing calls are OK. INVITE, unauthorized, INVITE with password, OK, RINGING,... Troubles with incoming calls / incoming INVITE's . I can not identify endpoint by IP, I have multiple registrations on the same Server1. As far as I understood, res_pjsip_endpoint_identifier_user match endpoint by "From" header, so it will not match also. match_headers also seems useless (not able to match "INVITE" string, just headers like "TO:"). Is there any way to match incoming INVITE calls ? (also OPTIONS, NOTIFY, ... packets) It should be a typical scenario, but it does not work... Is there any way to make it working ? [0.0.0.0-udp] type=transport protocol=udp bind=0.0.0.0:5060 [endpoint0](!) type=endpoint transport=0.0.0.0-udp disallow=all allow=alaw allow=ulaw t38_udptl=no t38_udptl_ec=none fax_detect=no t38_udptl_nat=no dtmf_mode=auto direct_media=yes from_domain=172.16.25.23 timers_sess_expires=1800 tone_zone=ru language=ru rewrite_contact=yes rtp_symmetric=yes force_rport=yes [registration0](!) type=registration transport=0.0.0.0-udp retry_interval=60 max_retries=10 expiration=3600 auth_rejection_permanent=yes server_uri=sip:172.16.25.23 [fxs17](endpoint0) context=from-sip-fxs aors=fxs17 outbound_auth=fxs17 from_user=1121 set_var=DAHDICHAN=17 [fxs17] type=aor qualify_frequency=60 contact=sip:1121@172.16.25.23 [fxs17] type=auth auth_type=userpass password=11 username=1121 [fxs17](registration0) outbound_auth=fxs17 client_uri=sip:1121@172.16.25.23 contact_user=fxs17 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming SIP call is rejected always.
Hi Have an asterisk. Setup a couple of friends. Sip.conf - http://pastebin.com/zUgiYbBi Trying to make incoming call, and have such error(cli output) http://pastebin.com/zFfgYcNR NOTICE[4994]: chan_sip.c:23316 handle_request_invite: Call from 'RMT20' (192.168.8.1:5062) to extension '4001020' rejected because extension not found in context 'rmt-context'. But, as you see, there is such extension. What I'm doing wrong? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP call is rejected always.
Maybe it needs to be _4001020? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yaroslav Panych Sent: Tuesday, April 17, 2012 7:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Incoming SIP call is rejected always. Hi Have an asterisk. Setup a couple of friends. Sip.conf - http://pastebin.com/zUgiYbBi Trying to make incoming call, and have such error(cli output) http://pastebin.com/zFfgYcNR NOTICE[4994]: chan_sip.c:23316 handle_request_invite: Call from 'RMT20' (192.168.8.1:5062) to extension '4001020' rejected because extension not found in context 'rmt-context'. But, as you see, there is such extension. What I'm doing wrong? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP call is rejected always.
2012/4/17 Danny Nicholas da...@debsinc.com: Maybe it needs to be _4001020? Not, it doesn't. Actually I have traced this incoming call step by step. Real reason it refuses - wrong domain. But why it wrong - have not any idea. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP call is rejected always.
Without knowing the URI the INVITE request was addressed to, its difficult to say what might be the actual cause of this. However, in your SIP configuration you have set allowexternaldomains to no. That implies that if the domain of the URI does not match any of the allowed domains you have set, that the INVITE request will be rejected. I imagine that this is the case, as ASTERISK-19601 noted that when this situation occurs, the NOTICE message indicates that there is a failure to match the extension, as opposed to a failure to match an allowed domain. Matthew Jordan Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org - Original Message - From: Yaroslav Panych panyc...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 17, 2012 4:58:14 PM Subject: Re: [asterisk-users] Incoming SIP call is rejected always. 2012/4/17 Danny Nicholas da...@debsinc.com: Maybe it needs to be _4001020? Not, it doesn't. Actually I have traced this incoming call step by step. Real reason it refuses - wrong domain. But why it wrong - have not any idea. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP call is rejected always.
2012/4/18 Matthew Jordan mjor...@digium.com: I imagine that this is the case, as ASTERISK-19601 noted that when this situation occurs, the NOTICE message indicates that there is a failure to match the extension, as opposed to a failure to match an allowed domain. Yes, it was hell to detect real error cause(I was forced to learn how to debug in KDevelop in less than four hours). Yes, it looks like ASTERISK-19601. But still I cannot understand why asterisk extracts wrong domain from request. However, in your SIP configuration you have set allowexternaldomains to no. Yes, it is intended. Without knowing the URI the INVITE request was addressed to, its difficult to say what might be the actual cause of this. I first letter I have provided CLI log which contains full request packets(Authless and authed INVITE included). Probably I do not understand how to configure Asterisk: I have one asterisk. It serves SIP domain example.com. This asterisk must be able to establish session with registered client of this account and also must be able to accept incoming sessions. No sessions with 3rd-party accounts on 3rd-party domains allowed to established. How I should setup this asterisk? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP call is rejected always.
- Original Message - From: Yaroslav Panych panyc...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 17, 2012 6:56:17 PM Subject: Re: [asterisk-users] Incoming SIP call is rejected always. 2012/4/18 Matthew Jordan mjor...@digium.com: I imagine that this is the case, as ASTERISK-19601 noted that when this situation occurs, the NOTICE message indicates that there is a failure to match the extension, as opposed to a failure to match an allowed domain. Yes, it was hell to detect real error cause(I was forced to learn how to debug in KDevelop in less than four hours). Yes, it looks like ASTERISK-19601. But still I cannot understand why asterisk extracts wrong domain from request. However, in your SIP configuration you have set allowexternaldomains to no. Yes, it is intended. Without knowing the URI the INVITE request was addressed to, its difficult to say what might be the actual cause of this. I first letter I have provided CLI log which contains full request packets(Authless and authed INVITE included). Probably I do not understand how to configure Asterisk: I have one asterisk. It serves SIP domain example.com. This asterisk must be able to establish session with registered client of this account and also must be able to accept incoming sessions. No sessions with 3rd-party accounts on 3rd-party domains allowed to established. How I should setup this asterisk? Well, I can't tell you how to configure your Asterisk server. However, I can tell you why Asterisk rejected the INVITE request. The URI that the INVITE request was addressed to is 4001020@192.168.8.2:5060. The domain portion of this URI is 192.168.8.2. Hence, the allowed domains need to include that particular IPv4 address. Looking at the allowed domains you've specified in sip.conf, we have: domain=sop-korniychuk domain=192.168.8.1 domain=192.168.8.1:5062 So, since the INVITE request does not match any of those three domains, its rejected. Note: I noticed that you have autodomain set to yes; I'm going to assume that the IPv4 address 192.168.8.2 is not associated with the server. Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [ Context 'default' created by 'pbx_config' ] 's' =1. Wait(1)[pbx_config] 2. Answer() [pbx_config] 3. Background(welcome)[pbx_config] 4. Background(and)[pbx_config] 5. Background(thank-you-for-calling) [pbx_config] 6. Background(conference-reservations)[pbx_config] 7. Waitfor() [pbx_config] 8. Hangup() [pbx_config] Unfortunately, no matter how I configure extensions.conf or sip.conf, the phone call always ends up saying: Extension is unavailable. Please leave your message after the tone. sip.conf: [general] register = NPANXX:passw...@service_provider_ip registertimeout=29 registerattempts=0 defaultexpiry=60 [DID_NUMBER] type=peer context=default host=SERVICE_PROVIDER_IP authuser=DID_NUMBER fromuser=DID_NUMBER fromdomain=SERVICE_PROVIDER_REALM remotesecret=SERVICE_PROVIDER_PASSWD secret=SERVICE_PROVIDER_PASSWD dtmfmode=rfc2833 disallow=all allow=ulaw qualify=yes I am attempting just to get the starting point where I can direct users through my asterisk box, but it won't direct users to the 's' extention, only to some voicemail box. I've removed the voicemail config. My extensions.conf is tiny: [globals] [general] [default] exten = s,1,Wait(1) exten = s,n,Answer() exten = s,n,Background(welcome) exten = s,n,Background(and) exten = s,n,Background(thank-you-for-calling) exten = s,n,Background(conference-reservations) exten = s,n,Waitfor() exten = s,n,Hangup() What am I doing wrong here? Thanks for any help you can give. Joe -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
On Wed, Aug 4, 2010 at 8:52 PM, Joe Wood sch...@gmail.com wrote: Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [ Context 'default' created by 'pbx_config' ] 's' =1. Wait(1) [pbx_config] 2. Answer() [pbx_config] 3. Background(welcome) [pbx_config] 4. Background(and) [pbx_config] 5. Background(thank-you-for-calling) [pbx_config] 6. Background(conference-reservations) [pbx_config] 7. Waitfor() [pbx_config] 8. Hangup() [pbx_config] Unfortunately, no matter how I configure extensions.conf or sip.conf, the phone call always ends up saying: Extension is unavailable. Please leave your message after the tone. sip.conf: [general] register = NPANXX:passw...@service_provider_ip registertimeout=29 registerattempts=0 defaultexpiry=60 [DID_NUMBER] type=peer context=default host=SERVICE_PROVIDER_IP authuser=DID_NUMBER fromuser=DID_NUMBER fromdomain=SERVICE_PROVIDER_REALM remotesecret=SERVICE_PROVIDER_PASSWD secret=SERVICE_PROVIDER_PASSWD dtmfmode=rfc2833 disallow=all allow=ulaw qualify=yes I am attempting just to get the starting point where I can direct users through my asterisk box, but it won't direct users to the 's' extention, only to some voicemail box. I've removed the voicemail config. My extensions.conf is tiny: [globals] [general] [default] exten = s,1,Wait(1) exten = s,n,Answer() exten = s,n,Background(welcome) exten = s,n,Background(and) exten = s,n,Background(thank-you-for-calling) exten = s,n,Background(conference-reservations) exten = s,n,Waitfor() exten = s,n,Hangup() What am I doing wrong here? Thanks for any help you can give. Joe You don't have any extensions in your default context that match the extension that your sip peer is dialing in on. 's' is not a default extension for SIP...try using _X., and see what you get. Bump up the CLI (core set verbose 10) and then repost a failed called attempt. Some SIP providers also use a + symbol in front of their inbound calls, so you may need to use _+X., instead. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
I don't see any On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby wcse...@selbytech.com wrote: You don't have any extensions in your default context that match the extension that your sip peer is dialing in on. 's' is not a default extension for SIP...try using _X., and see what you get. Bump up the CLI (core set verbose 10) and then repost a failed called attempt. Some SIP providers also use a + symbol in front of their inbound calls, so you may need to use _+X., instead. I don't see any call attempt/logs when I bump up the verbosity, and when I check my verbose logs I show: [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Registered extension context 'default' (0xb77980c0) in local table 0xb77960c0; registrar: pbx_config [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 1 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 2 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 3 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 4 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 5 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 6 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 7 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 8 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Registered extension context 'parkedcalls' (0xb7797ee0) in local table 0xb77960c0; registrar: features [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- merging incls/swits/igpats from old(parkedcalls) to new(parkedcalls) context, registrar = pbx_config [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '700' priority 1 to parkedcalls (0xb7797ee0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Time to scan old dialplan and merge leftovers back into the new: 0.89 sec [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Time to restore hints and swap in new dialplan: 0.02 sec [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Time to delete the old dialplan: 0.11 sec [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Total time merge_contexts_delete: 0.000102 sec [Aug 4 19:17:04] VERBOSE[12255] netsock.c: == Using SIP RTP CoS mark 5 [Aug 4 19:19:04] VERBOSE[12255] netsock.c: == Using SIP RTP CoS mark 5 [Aug 4 19:21:39] VERBOSE[12255] netsock.c: == Using SIP RTP CoS mark 5 I get the same error. Same random voicemail when no voicemail is configured. I was under the impressing that s was the catchall for all incoming trunks. What has changed? Joe -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood sch...@gmail.com wrote: I don't see any On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby wcse...@selbytech.com wrote: You don't have any extensions in your default context that match the extension that your sip peer is dialing in on. 's' is not a default extension for SIP...try using _X., and see what you get. Bump up the CLI (core set verbose 10) and then repost a failed called attempt. Some SIP providers also use a + symbol in front of their inbound calls, so you may need to use _+X., instead. I don't see any call attempt/logs when I bump up the verbosity, and when I check my verbose logs I show: The next step would be to enable sip debug on the peer you're trying to receive calls from (sip set debug peer PEERNAME or sip set debug ip IPADDRESS). Then send another call inbound and see what's happening. As far as the 's' extension, that's the start extension, it's used when no other extension information is presented. Pretty much every SIP peer I've ever seen presents an extension when entering a context, and thus the 's' extension doesn't catch it. I've typically only seen 's' used in Macros and with inbound analog lines. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
On Wed, Aug 4, 2010 at 7:49 PM, Warren Selby wcse...@selbytech.com wrote: On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood sch...@gmail.com wrote: I don't see any On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby wcse...@selbytech.com wrote: You don't have any extensions in your default context that match the extension that your sip peer is dialing in on. 's' is not a default extension for SIP...try using _X., and see what you get. Bump up the CLI (core set verbose 10) and then repost a failed called attempt. Some SIP providers also use a + symbol in front of their inbound calls, so you may need to use _+X., instead. I don't see any call attempt/logs when I bump up the verbosity, and when I check my verbose logs I show: The next step would be to enable sip debug on the peer you're trying to receive calls from (sip set debug peer PEERNAME or sip set debug ip IPADDRESS). Then send another call inbound and see what's happening. As far as the 's' extension, that's the start extension, it's used when no other extension information is presented. Pretty much every SIP peer I've ever seen presents an extension when entering a context, and thus the 's' extension doesn't catch it. I've typically only seen 's' used in Macros and with inbound analog lines. My experience with Asterisk in the past has been with inbound analog lines so that would make sense :) See if you spot anything weird here: --- SIP read from UDP:209.221.186.98:5060 --- INVITE sip:s...@209.221.186.50 SIP/2.0 Record-Route: sip:209.221.186.98;ftag=f7093e2d7e16a927d0816f6f5ed7aba4;lr Via: SIP/2.0/UDP 209.221.186.98;branch=z9hG4bK6b95.ae84c9620e19761029084a0f85255c29.0 Via: SIP/2.0/UDP 209.221.186.98:5071;branch=z9hG4bKf0a046579186bf058ba7783ed98a5a66;rport=5071 Max-Forwards: 16 From: 2538544199 sip:2538544...@209.221.186.98;tag=f7093e2d7e16a927d0816f6f5ed7aba4 To: sip:2063161...@209.221.186.98 Call-ID: ba711218-1ae3-122e-41bd-18a905721e58-b2b_1 CSeq: 200 INVITE Contact: Anonymous sip:2538544...@209.221.186.98:5071 Expires: 300 User-Agent: Sippy Softswitch v2.0.80 cisco-GUID: 1225641884-3786690633-966044271-4144140181 h323-conf-id: 1225641884-3786690633-966044271-4144140181 Content-Length: 321 Content-Type: application/sdp v=0 o=- 1280279699622 1280279699622 IN IP4 209.221.186.98 s=- c=IN IP4 209.221.186.98 t=0 0 m=audio 60304 RTP/AVP 0 a=fmtp:4 bitrate=6300;annexa=no a=rtpmap:96 iLBC/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=fmtp:18 annexb=no a=rtpmap:98 telephone-event/8000 a=oldmediaip:208.76.155.20 a=nortpproxy:yes - [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 0 [ 35]: INVITE sip:s...@209.221.186.50 SIP/2.0 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 1 [ 75]: Record-Route: sip:209.221.186.98;ftag=f7093e2d7e16a927d0816f6f5ed7aba4;lr [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 2 [ 85]: Via: SIP/2.0/UDP 209.221.186.98;branch=z9hG4bK6b95.ae84c9620e19761029084a0f85255c29.0 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 3 [ 94]: Via: SIP/2.0/UDP 209.221.186.98:5071;branch=z9hG4bKf0a046579186bf058ba7783ed98a5a66;rport=5071 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 4 [ 16]: Max-Forwards: 16 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 5 [ 85]: From: 2538544199 sip:2538544...@209.221.186.98;tag=f7093e2d7e16a927d0816f6f5ed7aba4 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 6 [ 35]: To: sip:2063161...@209.221.186.98 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 7 [ 51]: Call-ID: ba711218-1ae3-122e-41bd-18a905721e58-b2b_1 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 8 [ 16]: CSeq: 200 INVITE [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 9 [ 55]: Contact: Anonymous sip:2538544...@209.221.186.98:5071 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 10 [ 12]: Expires: 300 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 11 [ 36]: User-Agent: Sippy Softswitch v2.0.80 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 12 [ 54]: cisco-GUID: 1225641884-3786690633-966044271-4144140181 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 13 [ 56]: h323-conf-id: 1225641884-3786690633-966044271-4144140181 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 14 [ 19]: Content-Length: 321 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 15 [ 29]: Content-Type: application/sdp [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 16 [ 0]: [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:Body 0 [ 3]: v=0 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:Body 1 [ 53]: o=- 1280279699622 1280279699622 IN IP4 209.221.186.98 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:
Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
On Wed, Aug 4, 2010 at 10:25 PM, Joe Wood sch...@gmail.com wrote: On Wed, Aug 4, 2010 at 7:49 PM, Warren Selby wcse...@selbytech.com wrote: On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood sch...@gmail.com wrote: My experience with Asterisk in the past has been with inbound analog lines so that would make sense :) See if you spot anything weird here: Try adding insecure=invite to the DID_NUMBER peer, reload SIP and try your call again. By the way, it looks like your SIP provider has a built-in auto-failover to voicemail setup. You may want to get them to disable that once you get everything working on your end. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP and the 's' extension
Your ITSP is giving you the DNIS digits. You have to match them in your dialplan. What if your ITSP routed calls from FIFTY different numbers to your switch? How would you differetntiate between them if they all just routed to the S extension? That's why the paradigm is based on passing and matching digits. Otherwise you'd need fifty contexts/registrations. Number X routes to Jimmy, Y to Johnny, Z to customer service queue. Outbound calls starting with *67 block the outgoing caller ID. See what I mean? -Karl - Original Message - From: John A. Sullivan III jsulli...@opensourcedevel.com To: asterisk-users@lists.digium.com Sent: Wednesday, June 17, 2009 10:06 PM Subject: [asterisk-users] Incoming SIP and the 's' extension Hello, all. My apologies up front but I must be brain cramping on something very simple. I've tried to pare down my configuration to the absolute minimum for SIP traffic just to understand how it works. My incoming calls are not finding the s extension in my dial-plan. I am assuming SIP calls can do this. I am using Asterisk 1.6.1.1 sip.conf has nothing but: [general] context=incoming extensions.conf has: [globals] [general] autofallthrough=yes [default] ;exten = s,1,Verbose(1,Unrouted call handler) ;exten = s,n,Answer() ;exten = s,n,Wait(1) ;exten = s,n,Playback(tt-weasels) ;exten = s,n,Hangup() [incoming] exten = s,1,Answer() exten = s,n,Playback(hello-world) exten = s,n,Hangup() [internal] ;exten = 515,1,Verbose(1,Echo test application) ;exten = 515,1,Answer() ;exten = 515,n,Echo() ;exten = 515,n,Hangup() ;exten = 1000,1,Verbose(1,Extension 1000) ;exten = 1000,n,Dial(SIP/1000,30) ;exten = 1000,n,Hangup() ;exten = 1001,1,Verbose(1,Extension 1001) ;exten = 1001,n,Dial(SIP/1001,30) ;exten = 1001,n,Hangup() [phones] include = internal I then fire up twinkle on my desktop and dial sip:3...@pbx.mycompany.com. The Asterisk console shows: [Jun 17 22:58:55] NOTICE[32060]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '36' rejected because extension not found. If I use the same extensions.conf but change s to 36, it works. I would have expected the SIP channel to see that it had nothing which matched my name or IP address and sent processing to the [incoming] context where it would encounter s and process accordingly. What concept am I missing? Does SIP always have a FROM and TO and thus never uses s? I'm obviously misunderstanding a fundamental concept. Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Incoming SIP and the 's' extension
On Thu, 2009-06-18 at 03:50 +, Joseph L. Casale wrote: I then fire up twinkle on my desktop and dial sip:3...@pbx.mycompany.com. The Asterisk console shows: [Jun 17 22:58:55] NOTICE[32060]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '36' rejected because extension not found. If I use the same extensions.conf but change s to 36, it works. I would have expected the SIP channel to see that it had nothing which matched my name or IP address and sent processing to the [incoming] context where it would encounter s and process accordingly. http://www.voip-info.org/wiki/view/Asterisk+s+extension http://voip-info.ehd.com.br/wiki/view/Asterisk+standard+extensions.html What concept am I missing? Does SIP always have a FROM and TO and thus never uses s? I'm obviously misunderstanding a fundamental concept. Thanks - John You have a known #, your explicitly calling 36 from your soft phone. What you want is a pattern match for your sip phones, and the s for a dahdi line for example... snip Ah, ok. Thanks very much. That's what I thought might be happening but didn't trust my instincts over my ignorance and over the tutorials I was following which did not point that out when describing a minimal dialplan. It makes perfect sense. Thanks again - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Incoming SIP and the 's' extension
Hello, all. My apologies up front but I must be brain cramping on something very simple. I've tried to pare down my configuration to the absolute minimum for SIP traffic just to understand how it works. My incoming calls are not finding the s extension in my dial-plan. I am assuming SIP calls can do this. I am using Asterisk 1.6.1.1 sip.conf has nothing but: [general] context=incoming extensions.conf has: [globals] [general] autofallthrough=yes [default] ;exten = s,1,Verbose(1,Unrouted call handler) ;exten = s,n,Answer() ;exten = s,n,Wait(1) ;exten = s,n,Playback(tt-weasels) ;exten = s,n,Hangup() [incoming] exten = s,1,Answer() exten = s,n,Playback(hello-world) exten = s,n,Hangup() [internal] ;exten = 515,1,Verbose(1,Echo test application) ;exten = 515,1,Answer() ;exten = 515,n,Echo() ;exten = 515,n,Hangup() ;exten = 1000,1,Verbose(1,Extension 1000) ;exten = 1000,n,Dial(SIP/1000,30) ;exten = 1000,n,Hangup() ;exten = 1001,1,Verbose(1,Extension 1001) ;exten = 1001,n,Dial(SIP/1001,30) ;exten = 1001,n,Hangup() [phones] include = internal I then fire up twinkle on my desktop and dial sip:3...@pbx.mycompany.com. The Asterisk console shows: [Jun 17 22:58:55] NOTICE[32060]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '36' rejected because extension not found. If I use the same extensions.conf but change s to 36, it works. I would have expected the SIP channel to see that it had nothing which matched my name or IP address and sent processing to the [incoming] context where it would encounter s and process accordingly. What concept am I missing? Does SIP always have a FROM and TO and thus never uses s? I'm obviously misunderstanding a fundamental concept. Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Incoming SIP and the 's' extension
I then fire up twinkle on my desktop and dial sip:3...@pbx.mycompany.com. The Asterisk console shows: [Jun 17 22:58:55] NOTICE[32060]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '36' rejected because extension not found. If I use the same extensions.conf but change s to 36, it works. I would have expected the SIP channel to see that it had nothing which matched my name or IP address and sent processing to the [incoming] context where it would encounter s and process accordingly. http://www.voip-info.org/wiki/view/Asterisk+s+extension http://voip-info.ehd.com.br/wiki/view/Asterisk+standard+extensions.html What concept am I missing? Does SIP always have a FROM and TO and thus never uses s? I'm obviously misunderstanding a fundamental concept. Thanks - John You have a known #, your explicitly calling 36 from your soft phone. What you want is a pattern match for your sip phones, and the s for a dahdi line for example... jlc ___ -- 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] Incoming SIP call ring timeout
I had my incoming call time set 120 seconds before going to voicemail, apparently this timeout is longer than some existing timeout of ~60 seconds and the call terminates before it reaches my voicemail command. Is this an Asterisk default setting or could this be something on my SIP providers end? Thanks! jlc ___ -- 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] Incoming SIP callerid
Hi all,
I want to pass the incoming SIP callerid in Dial application:
Asterisk 1.2.13
sip.conf:
register = user:[EMAIL PROTECTED]/ext
extensions.conf:
exten = ext,1,Dial(SIP/phone1SIP/phone2)
on phone's display I see the 'ext' number, not the incoming SIP callerid as
can be seen on incoming calls when I register the phone directly to
provider.
I tried to add 'o' option to Dial but same results.
The closest thing (and perhaps not the smartest) I could do:
exten = ext,1,Set(CALLERID(name)=${SIPURI})
exten = ext,2,Dial(SIP/phone1SIP/phone2)
If that matters, phone1 and phone2 are Grandstream HTs with callerid-capable
cordless phones connected to FXS ports.
How can I get the calling number (and perhaps a hint to distinguish the SIP
account for incoming call) to be displayed on those phones?
Thank you in advance!
Best wishes,
Cristian
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RE: [asterisk-users] incoming SIP call
If your SIP server loses REGISTERs then it cant place an inbound SIP call. Try changing the REGISTER frequency to lower value. When you see incoming SIP call fail, you might want to check whether the REGISTERs are working. Thanks, Neel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean Marc Le Fevre Sent: Wednesday, April 18, 2007 11:15 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] incoming SIP call Hello all, I'm having a quite simple configuration like: SIP provider = asterisk SIP = lan Everythings works fine but sometime I can't get incoming call. here are some of the logs from set debug 25 set verbosity 25 sip show debug and sip.conf and a part of extension.conf thanks in advance Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;branch= z9hG4bK67c2df66;rport From: asterisk sip:[EMAIL PROTECTED];tag=as01265eaf To: sip:freephonie.net Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 18 Apr 2007 13:57:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 12 headers, 0 lines Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;b ranch=z9hG4bK253c1a3d;rport From: asterisk sip:[EMAIL PROTECTED];tag=as372da2cb To: sip:freephonie.net Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 18 Apr 2007 13:57:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Zpro*CLI -- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Call-ID: 7263e88c20c9f3 mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asterisk sip:[EMAIL PROTECTED];tag=as01265eaf To: sip:freephonie.net;tag=00-31057-001dc208-591e1ca81 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK67c2df6 6 Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' Zpro*CLI -- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asteris k sip:[EMAIL PROTECTED];tag=as372da2cb To: sip:freephonie.net;tag=00-32700-001dc209-6fc2b3303 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK253c1a3 d Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' sip.conf [general] context=incoming realm=etatcritik.dyndns.org bindport=5060 bindaddr=0.0.0.0 srvlookup=no maxexpiry=3600 defaultexpiry=1800 videosupport=yes disallow=all all ow=ulaw allow=ilbc allow=alaw allow=gsm musicclass=default language=fr useragent=Asterisk PBX dtmfmode = auto register = 09:[EMAIL PROTECTED] registertimeout=40 externip = 82.XXX.XXX.XXX localnet=10.XXX.XXX.XXX/255.255.255.0 qualify=6 nat = yes [test] type=friend username=test secret=test host=dynamic context=home callerid =test dmtfmode=rfc2833 authuser=test fromuser=test allow=all [freephonie_outbound] type=peer allow=all host=freephonie.net secret=SECRET fromuser=09XXX username=09XXX dtmfmode=inband qualify=6 fromdomain=freephonie.net [freep honie_inbound] type=peer context=incoming host=freephonie.net qualify=6 allow=all deny=0.0.0.0/0..0.0.0 permit=212.27.52.5/255.255.255.255 ; ip de freephonie.net etension.conf ... [incoming] exten = s,1,Ringing exten = s,2,Noop(I receive a sip call); exten = s,n,Goto(home,1000,1) exten = s,n,Congestion ; ... !DSPAM:462643f450705772331342! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming SIP call
Well thanks for answering, When I test, I use my GSM and call the number my provider gives me. How often it works or not, I didn't make test like 10 calls per hour for a pretty long time so I can't exactly tell. When I test, well sometimes it works great, sometime, the incoming call is redirected to an phone that is connected on my DSL box. I didn't see the error message SIP/2.0 403 not registered, but in that case: 1) I can make a call from asterisk to a gsm call (so It goes IAX phone = asterisk = SIP provider = GSM. 2) if I do show sip register in asterisk CLI, I can see I'm registered (or I may be misinterpretting this command. What can I do to investigate this registration message ? Is there an special debug command ? thanks :) From: Jean Marc Le Fevre [EMAIL PROTECTED] Date: Wed, 18 Apr 2007 18:14:41 +0200 Hello all, I'm having a quite simple configuration like: SIP provider = asterisk SIP = lan Everythings works fine but sometime I can't get incoming call. Define sometimes and from where the income call you can't get? here are some of the logs from set debug 25 set verbosity 25 sip show debug and sip.conf and a part of extension.conf thanks in advance [good stuff sniffed] Where do you suspect the error message is? --- Zpro*CLI -- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Does this message make sense, not registered? Yuan Liu Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asterisk sip:[EMAIL PROTECTED];tag=as01265eaf To: sip:freephonie.net;tag=00-31057-001dc208-591e1ca81 Via: SIP/2.0/UDP 82.XXX.XXX.XXX: 5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK67c2df66 Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' Zpro*CLI -- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asterisk sip:[EMAIL PROTECTED];tag=as372da2cb To: sip:freephonie.net;tag=00-32700-001dc209-6fc2b3303 Via: SIP/2.0/UDP 82.XXX.XXX.XXX: 5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK253c1a3d Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' sip.conf [general] context=incoming realm=etatcritik.dyndns.org bindport=5060 bindaddr=0.0.0.0 srvlookup=no maxexpiry=3600 defaultexpiry=1800 videosupport=yes disallow=all allow=ulaw allow=ilbc allow=alaw allow=gsm musicclass=default language=fr useragent=Asterisk PBX dtmfmode = auto register = 09:[EMAIL PROTECTED] registertimeout=40 externip = 82.XXX.XXX.XXX localnet=10.XXX.XXX.XXX/255.255.255.0 qualify=6 nat = yes [test] type=friend username=test secret=test host=dynamic context=home callerid =test dmtfmode=rfc2833 authuser=test fromuser=test allow=all [freephonie_outbound] type=peer allow=all host=freephonie.net secret=SECRET fromuser=09XXX username=09XXX dtmfmode=inband qualify=6 fromdomain=freephonie.net [freephonie_inbound] type=peer context=incoming host=freephonie.net qualify=6 allow=all deny=0.0.0.0/0.0.0.0 permit=212.27.52.5/255.255.255.255 ; ip de freephonie.net etension.conf ... [incoming] exten = s,1,Ringing exten = s,2,Noop(I receive a sip call); exten = s,n,Goto(home,1000,1) exten = s,n,Congestion ; ... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:4627b6c350701639315548! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming SIP call
Hello and thanks for answering, As I just answer to Yuan LIU, what I don't understand, is that I can place an outbound call from asterisk to a gsm at the same time I can't get asterisk thought a inbound call. But I'll try what you advice me. I'll tell you the result of it Jean-Marc LE FEVRE Le 19 avr. 07 à 20:08, Bala Neelakantan a écrit : If your SIP server loses REGISTERs then it cant place an inbound SIP call. Try changing the REGISTER frequency to lower value. When you see incoming SIP call fail, you might want to check whether the REGISTERs are working. Thanks, Neel -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] On Behalf Of Jean Marc Le Fevre Sent: Wednesday, April 18, 2007 11:15 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] incoming SIP call Hello all, I'm having a quite simple configuration like: SIP provider = asterisk SIP = lan Everythings works fine but sometime I can't get incoming call. here are some of the logs from set debug 25 set verbosity 25 sip show debug and sip.conf and a part of extension.conf thanks in advance Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;branch= z9hG4bK67c2df66;rport From: asterisk sip:[EMAIL PROTECTED];tag=as01265eaf To: sip:freephonie.net Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 18 Apr 2007 13:57:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 12 headers, 0 lines Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;b ranch=z9hG4bK253c1a3d;rport From: asterisk sip:[EMAIL PROTECTED];tag=as372da2cb To: sip:freephonie.net Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 18 Apr 2007 13:57:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Zpro*CLI -- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Call-ID: 7263e88c20c9f3 [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asterisk sip:[EMAIL PROTECTED];tag=as01265eaf To: sip:freephonie.net;tag=00-31057-001dc208-591e1ca81 Via: SIP/2.0/UDP 82.XXX.XXX.XXX: 5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK67c2df66 Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' Zpro*CLI -- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asteris k sip:[EMAIL PROTECTED];tag=as372da2cb To: sip:freephonie.net;tag=00-32700-001dc209-6fc2b3303 Via: SIP/2.0/UDP 82.XXX.XXX.XXX: 5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK253c1a3d Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' sip.conf [general] context=incoming realm=etatcritik.dyndns.org bindport=5060 bindaddr=0.0.0.0 srvlookup=no maxexpiry=3600 defaultexpiry=1800 videosupport=yes disallow=all all ow=ulaw allow=ilbc allow=alaw allow=gsm musicclass=default language=fr useragent=Asterisk PBX dtmfmode = auto register = 09:[EMAIL PROTECTED] registertimeout=40 externip = 82.XXX.XXX.XXX localnet=10.XXX.XXX.XXX/255.255.255.0 qualify=6 nat = yes [test] type=friend username=test secret=test host=dynamic context=home callerid =test dmtfmode=rfc2833 authuser=test fromuser=test allow=all [freephonie_outbound] type=peer allow=all host=freephonie.net secret=SECRET fromuser=09XXX username=09XXX dtmfmode=inband qualify=6 fromdomain=freephonie.net [freep honie_inbound] type=peer context=incoming host=freephonie.net qualify=6 allow=all deny=0.0.0.0/0..0.0.0 permit=212.27.52.5/255.255.255.255 ; ip de freephonie.net etension.conf ... [incoming] exten = s,1,Ringing exten = s,2,Noop(I receive a sip call); exten = s,n,Goto(home,1000,1) exten = s,n,Congestion ; ... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:4627b30550701698699180! !DSPAM:4627b7bb50703422486060! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] incoming SIP call
Well, for outbound calls, the SIP Server challenges the INVITE with 401/407. Then Re-INVITE is sent which explains why outgoing call works. It is possible that the SIP Server doesnt check to see whether the caller is Registered. For inbound call, the SIP server needs to know the gateway contact information, and it is obtained only through REGISTER (if not statically configured in the SIP server, which is very unlikely). Thanks, Neel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean Marc Le Fevre Sent: Thursday, April 19, 2007 1:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] incoming SIP call Hello and thanks for answering, As I just answer to Yuan LIU, what I don't understand, is that I can place an outbound call from asterisk to a gsm at the same time I can't get asterisk thought a inbound call. But I'll try what you advice me. I'll tell you the result of it Jean-Marc LE FEVRE Le 19 avr. 07 à 20:08, Bala Neelakantan a écrit : If your SIP server loses REGISTERs then it cant place an inbound SIP call. Try changing the REGI STER frequency to lower value. When you see incoming SIP call fail, you might want to check whether the REGISTERs are working. Thanks, Neel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean Marc Le Fevre Sent: Wednesday, April 18, 2007 11:15 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] incoming SIP call Hello all, I'm having a quite simple configuration like: SIP provider = asterisk SIP = lan Everythings works fine but sometime I can't get incoming call. here are some of the logs from set debug 25 set verbosity 25 sip show debug and sip.conf and a part of extension.conf thanks in advance Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;branch= z9hG4bK67c2df66;rport From: asterisk sip:[EMAIL PROTECTED];tag=as01265eaf To: lt;sip:freephonie.net Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 18 Apr 2007 13:57:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 12 headers, 0 lines Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;b ranch=z9hG4bK253c1a3d;rport From: asterisk sip:[EMAIL PROTECTED];tag=as372da2cb To: sip:freephonie.net Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 18 Apr 2007 13:57:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Zpro*CLI -- S IP read from 212.27.52.5:5060: SIP/2.0 403 not registered Call-ID: 7263e88c20c9f3 mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asterisk sip:[EMAIL PROTECTED];tag=as01265eaf To: sip:freephonie.net;tag=00-31057-001dc 208-591e1ca81 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK67c2df6 6 Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ' Zpro*CLI -- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asteris k sip:[EMAIL PROTECTED];tag=as372da2cb To: sip:freephonie.net;tag=00-32700-001dc209-6fc2b3303 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK253c1a3 d Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' sip.conf [general] context=incoming realm=etatcritik.dyndns.org bindport=5060 bindaddr=0.0.0.0 srvlookup=no maxexpiry=3600 defaultexpiry=1800 videosupport=yes disallow=all all ow=ulaw allow=ilbc allow=alaw allow=gsm musicclass=default language=fr useragent=Asterisk PBX dtmfmode = auto register = 09:[EMAIL PROTECTED] registertimeout=40 externip = 82.XXX.XXX.XXX localnet=10.XXX.XXX.XXX/255.255.255.0 qualify=6 nat = yes [test] type=friend username=test secret=test host=dynamic context=home callerid =test dmtfmode=rfc2833 authuser=test fromuser=test allow=all [freephonie_outbound] type=peer allow=all host=freephonie.net secret=SECRET fromuser=09XXX username=09XXX dtmfmode=inband quali fy=6 fromdomain=freephonie.net [freep honie_inbound] type=peer context=incoming host=freephonie.net qualify=6 allow=all /P deny=0.0.0.0/0..0.0.0 permit=212.27.52.5/255.255.255.255 ; ip de freephonie.net etension.conf ... [incoming] exten = s,1,Ringing exten = s,2,Noop(I receive a sip call); exten = s,n,Goto(home,1000,1) exten = s,n,Congestion
[asterisk-users] incoming SIP call
Hello all, I'm having a quite simple configuration like: SIP provider = asterisk SIP = lan Everythings works fine but sometime I can't get incoming call. here are some of the logs from set debug 25 set verbosity 25 sip show debug and sip.conf and a part of extension.conf thanks in advance Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;branch=z9hG4bK67c2df66;rport From: asterisk sip:[EMAIL PROTECTED];tag=as01265eaf To: sip:freephonie.net Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 18 Apr 2007 13:57:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 12 headers, 0 lines Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;branch=z9hG4bK253c1a3d;rport From: asterisk sip:[EMAIL PROTECTED];tag=as372da2cb To: sip:freephonie.net Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 18 Apr 2007 13:57:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Zpro*CLI -- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asterisk sip:[EMAIL PROTECTED];tag=as01265eaf To: sip:freephonie.net;tag=00-31057-001dc208-591e1ca81 Via: SIP/2.0/UDP 82.XXX.XXX.XXX: 5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK67c2df66 Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' Zpro*CLI -- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asterisk sip:[EMAIL PROTECTED];tag=as372da2cb To: sip:freephonie.net;tag=00-32700-001dc209-6fc2b3303 Via: SIP/2.0/UDP 82.XXX.XXX.XXX: 5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK253c1a3d Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' sip.conf [general] context=incoming realm=etatcritik.dyndns.org bindport=5060 bindaddr=0.0.0.0 srvlookup=no maxexpiry=3600 defaultexpiry=1800 videosupport=yes disallow=all allow=ulaw allow=ilbc allow=alaw allow=gsm musicclass=default language=fr useragent=Asterisk PBX dtmfmode = auto register = 09:[EMAIL PROTECTED] registertimeout=40 externip = 82.XXX.XXX.XXX localnet=10.XXX.XXX.XXX/255.255.255.0 qualify=6 nat = yes [test] type=friend username=test secret=test host=dynamic context=home callerid =test dmtfmode=rfc2833 authuser=test fromuser=test allow=all [freephonie_outbound] type=peer allow=all host=freephonie.net secret=SECRET fromuser=09XXX username=09XXX dtmfmode=inband qualify=6 fromdomain=freephonie.net [freephonie_inbound] type=peer context=incoming host=freephonie.net qualify=6 allow=all deny=0.0.0.0/0.0.0.0 permit=212.27.52.5/255.255.255.255 ; ip de freephonie.net etension.conf ... [incoming] exten = s,1,Ringing exten = s,2,Noop(I receive a sip call); exten = s,n,Goto(home,1000,1) exten = s,n,Congestion ; ... !DSPAM:462643f450705772331342! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] incoming SIP call
From: Jean Marc Le Fevre [EMAIL PROTECTED] Date: Wed, 18 Apr 2007 18:14:41 +0200 Hello all, I'm having a quite simple configuration like: SIP provider = asterisk SIP = lan Everythings works fine but sometime I can't get incoming call. Define sometimes and from where the income call you can't get? here are some of the logs from set debug 25 set verbosity 25 sip show debug and sip.conf and a part of extension.conf thanks in advance [good stuff sniffed] Where do you suspect the error message is? --- Zpro*CLI -- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Does this message make sense, not registered? Yuan Liu Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asterisk sip:[EMAIL PROTECTED];tag=as01265eaf To: sip:freephonie.net;tag=00-31057-001dc208-591e1ca81 Via: SIP/2.0/UDP 82.XXX.XXX.XXX: 5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK67c2df66 Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' Zpro*CLI -- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asterisk sip:[EMAIL PROTECTED];tag=as372da2cb To: sip:freephonie.net;tag=00-32700-001dc209-6fc2b3303 Via: SIP/2.0/UDP 82.XXX.XXX.XXX: 5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK253c1a3d Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' sip.conf [general] context=incoming realm=etatcritik.dyndns.org bindport=5060 bindaddr=0.0.0.0 srvlookup=no maxexpiry=3600 defaultexpiry=1800 videosupport=yes disallow=all allow=ulaw allow=ilbc allow=alaw allow=gsm musicclass=default language=fr useragent=Asterisk PBX dtmfmode = auto register = 09:[EMAIL PROTECTED] registertimeout=40 externip = 82.XXX.XXX.XXX localnet=10.XXX.XXX.XXX/255.255.255.0 qualify=6 nat = yes [test] type=friend username=test secret=test host=dynamic context=home callerid =test dmtfmode=rfc2833 authuser=test fromuser=test allow=all [freephonie_outbound] type=peer allow=all host=freephonie.net secret=SECRET fromuser=09XXX username=09XXX dtmfmode=inband qualify=6 fromdomain=freephonie.net [freephonie_inbound] type=peer context=incoming host=freephonie.net qualify=6 allow=all deny=0.0.0.0/0.0.0.0 permit=212.27.52.5/255.255.255.255 ; ip de freephonie.net etension.conf ... [incoming] exten = s,1,Ringing exten = s,2,Noop(I receive a sip call); exten = s,n,Goto(home,1000,1) exten = s,n,Congestion ; ... !DSPAM:462643f450705772331342! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP line does not display CallerID correctly
Lee Jenkins wrote:
Hi all,
I've just setup a sip line with Telasip and when they route the calls to
my asterisk box, they include an extension along with the context that
is defined in sip.conf for that DID.
At first, I couldn't figure why they were getting 404 error from my
asterisk box, but then figured out that they are sending the call to an
extension that matches my number with them, in the context defined in
sip.conf. So instead of transferring the call to say, incoming they
are sending the call to incoming/55 where 55 is the
sip phone number I have with them.
So, I have to account for that extension with something like this:
[incoming]
exten=55,Goto(incoming,s,1)
Thus transferring the call to the context that I want it to come in on.
The problem that I have is the caller ID ${CALLERID(num)} always shows
the actual number provided by Telasip and not the actual caller id
information.
I also have axVoice and they do not do it this way. They simply send it
to the context without specifying an extension.
Below is a sip packet. The Caller ID comes through correctly on the sip
packet by for some reason as I mentioned, Asterisk is reporting it as
the number I have with the sip provider.
Below is the sip packet. The 33 represents my cell phone I
was using to call into the system, which was correct.
localhost*CLI exit
-- SIP read from 4.79.19.56:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Record-Route: sip:4.79.19.56;ftag=as5bfea671;lr=on
Via: SIP/2.0/UDP 4.79.19.56:5060;branch=0
Via: SIP/2.0/UDP
4.79.19.58:5060;received=192.168.3.5;branch=z9hG4bK3f2e414e;rport=5060
From: 33 sip:[EMAIL PROTECTED];tag=as5bfea671
To: sip:[EMAIL PROTECTED];tag=as12b47a8d
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Telasip GW3
Max-Forwards: 69
Remote-Party-ID: 33
sip:[EMAIL PROTECTED];privacy=off;screen=no
Content-Length: 0
P-hint: proxy loose route
Just a bit more information on this in hopes that someone more
knowledgeable than I will chime in.
This problem still persists, of course and I'm having a pickle trying to
figure out what the problem would be.
Summary:
I have two SIP accounts. One with axVoice who forwards the call to my
asterisk box at the s extension of my incoming context. The other
with Telasip who forwards the calls to 55 extension of my
incoming context where 55 represents the provisioned telephone
number for that line.
The problem is that with calls coming in from the Telasip account,
Asterisk ${CALLERID(num)} always returns 55, the Telasip
provisioned phone number instead of the caller's real CID.
My incoming extension.conf:
; -- Before adding Telasip account
[incoming]
exten=s,1,Answer()
exten=s,2,Ringing()
exten=s,3,SetMusicOnHold(default)
exten=s,4,Wait(1)
exten=s,5,Goto(check_time,s,1)
... check_time does a quick GoToIfTime and into the dialplan we go.
When I got my Telasip account, I had to change it to match the extension
they were pointing to
[incoming]
exten=s,1,Answer()
exten=s,2,Ringing()
exten=s,3,SetMusicOnHold(default)
exten=s,4,Wait(5)
exten=s,5,Goto(check_time,s,1)
exten=55,1,Answer()
exten=55,2,Ringing()
exten=55,3,SetMusicOnHold(default)
exten=55,4,Wait(5)
exten=55,5,Goto(check_time,s,1)
; exten=55,1,Goto(incoming,s,1 = Tried this too.
I've tried it a couple of different ways and while the calls do get
routed correctly and everything works fine otherwise, any reference to
${CALLERID(num)} returns the SIP number called and not the caller's
correct CID. For instance, inserting a Noop() such as ;
exten=55,2,Noop(caller id: ${CALLERID(num)})
results in caller id: 55 in the CLI when the call comes in.
Notice from the priority that I tried it after a call to Answer().
I'm a little stumped so any suggestions or observations would be
appreciated as always.
--
Warm Regards,
Lee
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[asterisk-users] Incoming SIP line does not display CallerID correctly
Hi all,
I've just setup a sip line with Telasip and when they route the calls to
my asterisk box, they include an extension along with the context that
is defined in sip.conf for that DID.
At first, I couldn't figure why they were getting 404 error from my
asterisk box, but then figured out that they are sending the call to an
extension that matches my number with them, in the context defined in
sip.conf. So instead of transferring the call to say, incoming they
are sending the call to incoming/55 where 55 is the
sip phone number I have with them.
So, I have to account for that extension with something like this:
[incoming]
exten=55,Goto(incoming,s,1)
Thus transferring the call to the context that I want it to come in on.
The problem that I have is the caller ID ${CALLERID(num)} always shows
the actual number provided by Telasip and not the actual caller id
information.
I also have axVoice and they do not do it this way. They simply send it
to the context without specifying an extension.
Below is a sip packet. The Caller ID comes through correctly on the sip
packet by for some reason as I mentioned, Asterisk is reporting it as
the number I have with the sip provider.
Below is the sip packet. The 33 represents my cell phone I
was using to call into the system, which was correct.
localhost*CLI exit
-- SIP read from 4.79.19.56:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Record-Route: sip:4.79.19.56;ftag=as5bfea671;lr=on
Via: SIP/2.0/UDP 4.79.19.56:5060;branch=0
Via: SIP/2.0/UDP
4.79.19.58:5060;received=192.168.3.5;branch=z9hG4bK3f2e414e;rport=5060
From: 33 sip:[EMAIL PROTECTED];tag=as5bfea671
To: sip:[EMAIL PROTECTED];tag=as12b47a8d
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Telasip GW3
Max-Forwards: 69
Remote-Party-ID: 33
sip:[EMAIL PROTECTED];privacy=off;screen=no
Content-Length: 0
P-hint: proxy loose route
--
Warm Regards,
Lee
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Re: [asterisk-users] Incoming sip line with INX (internationalnumber.com)
On 09/10/06, Daniel Cyt [EMAIL PROTECTED] wrote: Hi, I can't get my INX line working for incoming (outgoing is working fine). When I dial this number from my home phone, asterisk sends the call straight to extension 101, for some reason it doens't read what my extensions.conf is saying. SIP.CONF register = number:[EMAIL PROTECTED]/101 ;number is a replacement for my line number Didn't you think that the '101' there might be a clue? Your register statement tells the provider to deliver the call to '101'. Replace that with a different number and something different will happen - perhaps the rest of your dialplan is expecting the call to come in with a destination which matches your DID - in which case, put the DID number there instead of the 101. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming sip line with INX(internationalnumber.com)
Hi Peter, Thank you for your answer. I did: register = DID:[EMAIL PROTECTED]/DID exten = DID,1,... Now when I call the DID number It doesnt reach the Asterisk. sip show registry shows me the line is registered but when I dial out from my softphone (eyeBeam) I get the 500 error - disconnected and the message the person you are calling is unavailable. Please, what do you suggest me to do? From: Peter Bowyer [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Incoming sip line with INX(internationalnumber.com) Date: Mon, 9 Oct 2006 08:48:58 +0100 On 09/10/06, Daniel Cyt [EMAIL PROTECTED] wrote: Hi, I can't get my INX line working for incoming (outgoing is working fine). When I dial this number from my home phone, asterisk sends the call straight to extension 101, for some reason it doens't read what my extensions.conf is saying. SIP.CONF register = number:[EMAIL PROTECTED]/101 ;number is a replacement for my line number Didn't you think that the '101' there might be a clue? Your register statement tells the provider to deliver the call to '101'. Replace that with a different number and something different will happen - perhaps the rest of your dialplan is expecting the call to come in with a destination which matches your DID - in which case, put the DID number there instead of the 101. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming sip line with INX(internationalnumber.com)
On 09/10/06, Daniel Cyt [EMAIL PROTECTED] wrote: Hi Peter, Thank you for your answer. I did: register = DID:[EMAIL PROTECTED]/DID exten = DID,1,... Now when I call the DID number It doesnt reach the Asterisk. sip show registry shows me the line is registered but when I dial out from my softphone (eyeBeam) I get the 500 error - disconnected and the message the person you are calling is unavailable. Please, what do you suggest me to do? Have you matched up the 'context= ' entry for your SIP provider in sip.conf with the right context in extensions.conf where the 'exten = DID' is? Do a sip debug and see what it's telling you about the call, post it here if it doesn't help. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming sip line withINX(internationalnumber.com)
Ok, I've got asterisk stop and start over again and Its working!!! THANK YOU VERY MUCH From: Daniel Cyt [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Incoming sip line withINX(internationalnumber.com) Date: Mon, 09 Oct 2006 07:48:29 -0200 Hi Peter, Thank you for your answer. I did: register = DID:[EMAIL PROTECTED]/DID exten = DID,1,... Now when I call the DID number It doesnt reach the Asterisk. sip show registry shows me the line is registered but when I dial out from my softphone (eyeBeam) I get the 500 error - disconnected and the message the person you are calling is unavailable. Please, what do you suggest me to do? From: Peter Bowyer [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Incoming sip line with INX(internationalnumber.com) Date: Mon, 9 Oct 2006 08:48:58 +0100 On 09/10/06, Daniel Cyt [EMAIL PROTECTED] wrote: Hi, I can't get my INX line working for incoming (outgoing is working fine). When I dial this number from my home phone, asterisk sends the call straight to extension 101, for some reason it doens't read what my extensions.conf is saying. SIP.CONF register = number:[EMAIL PROTECTED]/101 ;number is a replacement for my line number Didn't you think that the '101' there might be a clue? Your register statement tells the provider to deliver the call to '101'. Replace that with a different number and something different will happen - perhaps the rest of your dialplan is expecting the call to come in with a destination which matches your DID - in which case, put the DID number there instead of the 101. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming sip line with INX (internationalnumber.com)
Hi, I can't get my INX line working for incoming (outgoing is working fine). When I dial this number from my home phone, asterisk sends the call straight to extension 101, for some reason it doens't read what my extensions.conf is saying. I had the same dial plan and same extensions.conf tested with a sip line from another provider and it was working good, so I guess the problem isn't in the dial plan. Also, my server hour is correct. WHAT I WANT: When asterisk answer the call and Im in business hours, the calls goes to a prompt, otherwise it would send the caller to the voicemail. WHAT I GOT: Asterisk doesn't read this, it send the call straight to extension 101. Could anybody please point me the directions to solve this problem? Thank you very much --- SIP.CONF register = number:[EMAIL PROTECTED]/101 ;number is a replacement for my line number [inx] type=peer username=number secret=password fromuser=number host=sip.intlno.com insecure=very disallow=all allow=g729 EXTENSIONS.CONF exten = number,1,GotoIfTime(08:00-16:59|*|*|*?4) exten = number,2,Goto(5) exten = number,3,Set(CALLERID(name)=USALine) exten = number,4,Goto(myprompt,s,1) exten = number,5,Background(danclosed) exten = number,6,Hangup() [myprompt] include = default exten = s,1,Ringing exten = s,2,Background(danprompt) exten = s,3,Wait(10) exten = s,4,Hangup() ;Option 1: english exten = 1,1,Set(CALLERID(name)=English) exten = 1,2,Dial(SIP/101,60,TtrA(danic)) exten = 1,3,Voicemail(su101) exten = 1,4,Hangup() _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming SIP provider goes unregistered and never recovers
Hello... We have a SIP provider for inbound dialtone that periodically goes into Unregistered state (per sip show registry) and doesn't seem to recover. Most often it seems to happen after storms, when our office DSL may have gone wacky for a little bit, but will then stay down even days later. Is this normal, and if so, is there a setting that can be set to have it retry? A simple reload fixes the problem, but it seems like it should recover itself. Best, JL. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Incoming SIP or IAX2 via NAT
Thanks for the replies guys (Chris Peter)... I think I've confused matters by not explaining things properly! My ISP has my internet connection on a private IP address - so my LAN has an address (192.168.42.*) and my internet connection has an address 10.100.x.x. That is then NAT'd again onto a single address used for a large majority of their customers. I won't go into the details of exactly why this is the case, but I used to work for them, and they're still not charging me for the connection, so I'm not complaining I am going to move off it, but it will be a few weeks before I can... so... Can IAX2 operate over double NAT? If so, what do I need to put in the register part of the config? Cheers Nunners -Original Message- From: Chris Bagnall [mailto:[EMAIL PROTECTED] Sent: 09 May 2006 22:21 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Incoming SIP or IAX2 via NAT Is there anyway in which I can successfully receive incoming calls from my Voip-Talk.org numbers (an 0845 number) without the static IP? Can you login to the voiptalk control panel and change the numbers to point at your current IP ? I've tried various options with SIP IAX2, but it would seem that neither will work. Has anyone got any suggestions? Not sure what you mean by that. Surely if your box was receiving calls from them before the IP change, and now isn't (on a different IP), then the only thing that needs changing should be the IP on their control panel. If that's not an option for whatever reason, your best bet might be to find someone with an asterisk box on a static IP in a datacentre to which you could get your voiptalk numbers pointed, and you could then have your box register periodically with that box. shameless plugas luck would have it, we have a few boxes that'd fit the bill, contact me off-list if you want to discuss it further/shameless plug Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming SIP or IAX2 via NAT
Ive installed successfully freePBX with Asterisk, and got various internal extensions working, however recently my internet facing IP address has been removed by my ISP (for various reason) and Im not going to be able to get it back for a few weeks. Is there anyway in which I can successfully receive incoming calls from my Voip-Talk.org numbers (an 0845 number) without the static IP? Ive tried various options with SIP IAX2, but it would seem that neither will work. Has anyone got any suggestions? Cheers Nunners ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Incoming SIP or IAX2 via NAT
Is there anyway in which I can successfully receive incoming calls from my Voip-Talk.org numbers (an 0845 number) without the static IP? Can you login to the voiptalk control panel and change the numbers to point at your current IP ? I've tried various options with SIP IAX2, but it would seem that neither will work. Has anyone got any suggestions? Not sure what you mean by that. Surely if your box was receiving calls from them before the IP change, and now isn't (on a different IP), then the only thing that needs changing should be the IP on their control panel. If that's not an option for whatever reason, your best bet might be to find someone with an asterisk box on a static IP in a datacentre to which you could get your voiptalk numbers pointed, and you could then have your box register periodically with that box. shameless plugas luck would have it, we have a few boxes that'd fit the bill, contact me off-list if you want to discuss it further/shameless plug Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming SIP or IAX2 via NAT
On 09/05/06, James Nunnerley [EMAIL PROTECTED] wrote: I've installed successfully freePBX with Asterisk, and got various internal extensions working, however… recently my internet facing IP address has been removed by my ISP (for various reason) and I'm not going to be able to get it back for a few weeks. Is there anyway in which I can successfully receive incoming calls from my Voip-Talk.org numbers (an 0845 number) without the static IP? I'm sure Voiptalk support would help you with this in not time at all, but... if you use the Voiptalk control panel you can route the DID to your Voiptalk ID hen 'register' to the Voiptalk ID from Asterisk. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming SIP Calls
I set up a deal with a voip provider to route calls to me via SIP. When
the call hits my system I get a busy signal. I have a route set up
through amp for the number (8002286573). Not sure what else I need to
set up. This is what I get at the CLI.
--
asterisk*CLI
-- Executing AbsoluteTimeout(SIP/71.16.179.175-0856d708, 15) in
new stack
-- Set Absolute Timeout to 15
-- Executing Congestion(SIP/71.16.179.175-0856d708, ) in new stack
== Spawn extension (from-sip-external, 8002286573, 2) exited non-zero
on 'SIP/71.16.179.175-0856d708'
-- Executing AbsoluteTimeout(SIP/71.16.179.175-0856d708, 15) in
new stack
-- Set Absolute Timeout to 15
-- Executing Congestion(SIP/71.16.179.175-0856d708, ) in new stack
== Spawn extension (from-sip-external, h, 2) exited non-zero on
'SIP/71.16.179.175-0856d708'
-- Executing AbsoluteTimeout(SIP/71.16.179.175-0856d708, 15) in
new stack
-- Set Absolute Timeout to 15
-- Executing Congestion(SIP/71.16.179.175-0856d708, ) in new stack
== Spawn extension (from-sip-external, 8002286573, 2) exited non-zero
on 'SIP/71.16.179.175-0856d708'
-- Executing AbsoluteTimeout(SIP/71.16.179.175-0856d708, 15) in
new stack
-- Set Absolute Timeout to 15
-- Executing Congestion(SIP/71.16.179.175-0856d708, ) in new stack
== Spawn extension (from-sip-external, h, 2) exited non-zero on
'SIP/71.16.179.175-0856d708'
-- Executing AbsoluteTimeout(SIP/71.16.179.175-0856d708, 15) in
new stack
-- Set Absolute Timeout to 15
-- Executing Congestion(SIP/71.16.179.175-0856d708, ) in new stack
== Spawn extension (from-sip-external, 8002286573, 2) exited non-zero
on 'SIP/71.16.179.175-0856d708'
-- Executing AbsoluteTimeout(SIP/71.16.179.175-0856d708, 15) in
new stack
-- Set Absolute Timeout to 15
-- Executing Congestion(SIP/71.16.179.175-0856d708, ) in new stack
== Spawn extension (from-sip-external, h, 2) exited non-zero on
'SIP/71.16.179.175-0856d708'
---
this is what I get in /var/log/asterisk/full
--
Jan 17 14:01:24 DEBUG[3033] chan_sip.c: Checking SIP call limits for device
Jan 17 14:01:24 DEBUG[3033] chan_sip.c: build_route: Contact hop:
sip:[EMAIL PROTECTED]:5060
Jan 17 14:01:24 VERBOSE[9282] logger.c: -- Executing
AbsoluteTimeout(SIP/71.16.179.175-0856ac50, 15) in new stack
Jan 17 14:01:24 VERBOSE[9282] logger.c: -- Set Absolute Timeout to 15
Jan 17 14:01:24 VERBOSE[9282] logger.c: -- Executing
Congestion(SIP/71.16.179.175-0856ac50, ) in new stack
Jan 17 14:01:24 VERBOSE[9282] logger.c: == Spawn extension
(from-sip-external, 8002286573, 2) exited non-zero on
'SIP/71.16.179.175-0856ac50'
Jan 17 14:01:24 VERBOSE[9282] logger.c: -- Executing
AbsoluteTimeout(SIP/71.16.179.175-0856ac50, 15) in new stack
Jan 17 14:01:24 VERBOSE[9282] logger.c: -- Set Absolute Timeout to 15
Jan 17 14:01:24 VERBOSE[9282] logger.c: -- Executing
Congestion(SIP/71.16.179.175-0856ac50, ) in new stack
Jan 17 14:01:24 VERBOSE[9282] logger.c: == Spawn extension
(from-sip-external, h, 2) exited non-zero on 'SIP/71.16.179.175-0856ac50'
Jan 17 14:01:24 DEBUG[9282] cdr_addon_mysql.c: cdr_mysql: inserting a
CDR record.
Jan 17 14:01:24 DEBUG[9282] cdr_addon_mysql.c: cdr_mysql: SQL command as
follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)
VALUES ('2006-01-17
14:01:24','6124322250','6124322250','8002286573','from-sip-external',
'SIP/71.16.179.175-0856ac50','','Congestion','',0,0,'NO
ANSWER',3,'','1137528084.126')
Jan 17 14:01:24 DEBUG[9282] chan_sip.c: update_call_counter() -
decrement call limit counter
Jan 17 14:01:24 DEBUG[3033] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 101:
Match Not Found
Jan 17 14:01:24 DEBUG[3033] chan_sip.c: Checking SIP call limits for device
Jan 17 14:01:24 DEBUG[3033] chan_sip.c: build_route: Contact hop:
sip:[EMAIL PROTECTED]:5060
Jan 17 14:01:24 VERBOSE[9286] logger.c: -- Executing
AbsoluteTimeout(SIP/71.16.179.175-0856ac50, 15) in new stack
Jan 17 14:01:24 VERBOSE[9286] logger.c: -- Set Absolute Timeout to 15
Jan 17 14:01:24 VERBOSE[9286] logger.c: -- Executing
Congestion(SIP/71.16.179.175-0856ac50, ) in new stack
Jan 17 14:01:24 VERBOSE[9286] logger.c: == Spawn extension
(from-sip-external, 8002286573, 2) exited non-zero on
'SIP/71.16.179.175-0856ac50'
Jan 17 14:01:24 VERBOSE[9286] logger.c: -- Executing
AbsoluteTimeout(SIP/71.16.179.175-0856ac50, 15) in new stack
Jan 17 14:01:24 VERBOSE[9286] logger.c: -- Set Absolute Timeout to 15
Jan 17 14:01:24 VERBOSE[9286] logger.c: -- Executing
Congestion(SIP/71.16.179.175-0856ac50, ) in new stack
Jan 17 14:01:24 VERBOSE[9286] logger.c: == Spawn extension
(from-sip-external, h, 2) exited non-zero on 'SIP/71.16.179.175-0856ac50'
Jan 17 14:01:24 DEBUG[9286] cdr_addon_mysql.c: cdr_mysql:
Re: [Asterisk-Users] Incoming SIP Calls
You are sending the call to from-sip-external which by default dumps
the call and gives the congestion message.
Go into your sip.conf and change from-sip-external to from-pstn or
change the context from-sip-external in extensions.conf to what you
want it to do.
My guess is you are using AAH.
On 1/17/06, Michael Sampson [EMAIL PROTECTED] wrote:
I set up a deal with a voip provider to route calls to me via SIP. When
the call hits my system I get a busy signal. I have a route set up
through amp for the number (8002286573). Not sure what else I need to
set up. This is what I get at the CLI.
--
asterisk*CLI
-- Executing AbsoluteTimeout(SIP/71.16.179.175-0856d708, 15) in
new stack
-- Set Absolute Timeout to 15
-- Executing Congestion(SIP/71.16.179.175-0856d708, ) in new stack
== Spawn extension (from-sip-external, 8002286573, 2) exited non-zero
on 'SIP/71.16.179.175-0856d708'
-- Executing AbsoluteTimeout(SIP/71.16.179.175-0856d708, 15) in
new stack
-- Set Absolute Timeout to 15
-- Executing Congestion(SIP/71.16.179.175-0856d708, ) in new stack
== Spawn extension (from-sip-external, h, 2) exited non-zero on
'SIP/71.16.179.175-0856d708'
-- Executing AbsoluteTimeout(SIP/71.16.179.175-0856d708, 15) in
new stack
-- Set Absolute Timeout to 15
-- Executing Congestion(SIP/71.16.179.175-0856d708, ) in new stack
== Spawn extension (from-sip-external, 8002286573, 2) exited non-zero
on 'SIP/71.16.179.175-0856d708'
-- Executing AbsoluteTimeout(SIP/71.16.179.175-0856d708, 15) in
new stack
-- Set Absolute Timeout to 15
-- Executing Congestion(SIP/71.16.179.175-0856d708, ) in new stack
== Spawn extension (from-sip-external, h, 2) exited non-zero on
'SIP/71.16.179.175-0856d708'
-- Executing AbsoluteTimeout(SIP/71.16.179.175-0856d708, 15) in
new stack
-- Set Absolute Timeout to 15
-- Executing Congestion(SIP/71.16.179.175-0856d708, ) in new stack
== Spawn extension (from-sip-external, 8002286573, 2) exited non-zero
on 'SIP/71.16.179.175-0856d708'
-- Executing AbsoluteTimeout(SIP/71.16.179.175-0856d708, 15) in
new stack
-- Set Absolute Timeout to 15
-- Executing Congestion(SIP/71.16.179.175-0856d708, ) in new stack
== Spawn extension (from-sip-external, h, 2) exited non-zero on
'SIP/71.16.179.175-0856d708'
---
this is what I get in /var/log/asterisk/full
--
Jan 17 14:01:24 DEBUG[3033] chan_sip.c: Checking SIP call limits for device
Jan 17 14:01:24 DEBUG[3033] chan_sip.c: build_route: Contact hop:
sip:[EMAIL PROTECTED]:5060
Jan 17 14:01:24 VERBOSE[9282] logger.c: -- Executing
AbsoluteTimeout(SIP/71.16.179.175-0856ac50, 15) in new stack
Jan 17 14:01:24 VERBOSE[9282] logger.c: -- Set Absolute Timeout to 15
Jan 17 14:01:24 VERBOSE[9282] logger.c: -- Executing
Congestion(SIP/71.16.179.175-0856ac50, ) in new stack
Jan 17 14:01:24 VERBOSE[9282] logger.c: == Spawn extension
(from-sip-external, 8002286573, 2) exited non-zero on
'SIP/71.16.179.175-0856ac50'
Jan 17 14:01:24 VERBOSE[9282] logger.c: -- Executing
AbsoluteTimeout(SIP/71.16.179.175-0856ac50, 15) in new stack
Jan 17 14:01:24 VERBOSE[9282] logger.c: -- Set Absolute Timeout to 15
Jan 17 14:01:24 VERBOSE[9282] logger.c: -- Executing
Congestion(SIP/71.16.179.175-0856ac50, ) in new stack
Jan 17 14:01:24 VERBOSE[9282] logger.c: == Spawn extension
(from-sip-external, h, 2) exited non-zero on 'SIP/71.16.179.175-0856ac50'
Jan 17 14:01:24 DEBUG[9282] cdr_addon_mysql.c: cdr_mysql: inserting a
CDR record.
Jan 17 14:01:24 DEBUG[9282] cdr_addon_mysql.c: cdr_mysql: SQL command as
follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)
VALUES ('2006-01-17
14:01:24','6124322250','6124322250','8002286573','from-sip-external',
'SIP/71.16.179.175-0856ac50','','Congestion','',0,0,'NO
ANSWER',3,'','1137528084.126')
Jan 17 14:01:24 DEBUG[9282] chan_sip.c: update_call_counter() -
decrement call limit counter
Jan 17 14:01:24 DEBUG[3033] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 101:
Match Not Found
Jan 17 14:01:24 DEBUG[3033] chan_sip.c: Checking SIP call limits for device
Jan 17 14:01:24 DEBUG[3033] chan_sip.c: build_route: Contact hop:
sip:[EMAIL PROTECTED]:5060
Jan 17 14:01:24 VERBOSE[9286] logger.c: -- Executing
AbsoluteTimeout(SIP/71.16.179.175-0856ac50, 15) in new stack
Jan 17 14:01:24 VERBOSE[9286] logger.c: -- Set Absolute Timeout to 15
Jan 17 14:01:24 VERBOSE[9286] logger.c: -- Executing
Congestion(SIP/71.16.179.175-0856ac50, ) in new stack
Jan 17 14:01:24 VERBOSE[9286] logger.c: == Spawn extension
(from-sip-external, 8002286573, 2) exited non-zero on
'SIP/71.16.179.175-0856ac50'
Jan 17 14:01:24 VERBOSE[9286] logger.c: -- Executing
[Asterisk-Users] Incoming SIP connection
Geetings to all. I am having a hell of a time getting incoming SIP connections to work properly, and am hoping that someone can help me. Here is what I am using as a guide (from the wiki): Incoming SIP Connections When Asterisk receives an incoming SIP call, the SIP Channel Module first tries to find a [user] section matching the caller name (From: username), then tries to find a [peer] section matching the caller's IP address. If no matching user or peer is found, the call is sent to the context defined in the [general] section of sip.conf. I am mainly concerned with the second point. I want to match an incoming SIP connection to a particular IP address. I have tried just about everything, and the connection always goes to the default context, or the context defined at the top of the sip.conf file. I would like to be able to direct incoming SIP connection to a particular set of extensions. There is no username and password involved as there will be many users coming from this one IP. This is what I have tried recently: [sipin_test] type=peer defaultip=195.27.242.120 context=test_trunk deny=0.0.0.0/0.0.0.0 permit=195.27.242.120/255.255.255.255 dtmfmode=rfc2833 disallow=all allow=ulaw nat=no I have also tried changing what is inside the brackets to the IP address. I have tried many many different combinations of the above, but the IP address never seems to get picked up correctly. I am testing the SIP connection using sipsak. I realize that Asterisk is probably not the best SIP server to use, and plan on migration to SER, but if anyone can offer any suggestions I would really appreciate it. Regards to all, Joe ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] Incoming SIP connection
Hi Joseph, Here's a basic entry for you that you should be able to adapt. [mypeer] Type=peer Host=ip or hostname Context=where to send the call Disallow=all Allow=ulaw Insecure=very The insecure=very causes Asterisk to not do any authentication and trust it based on the IP. Joshua Colp On 10/16/05 1:22 PM, Joseph Rothstein [EMAIL PROTECTED] wrote: Geetings to all. I am having a hell of a time getting incoming SIP connections to work properly, and am hoping that someone can help me. Here is what I am using as a guide (from the wiki): Incoming SIP Connections When Asterisk receives an incoming SIP call, the SIP Channel Module first tries to find a [user] section matching the caller name (From: username), then tries to find a [peer] section matching the caller's IP address. If no matching user or peer is found, the call is sent to the context defined in the [general] section of sip.conf. I am mainly concerned with the second point. I want to match an incoming SIP connection to a particular IP address. I have tried just about everything, and the connection always goes to the default context, or the context defined at the top of the sip.conf file. I would like to be able to direct incoming SIP connection to a particular set of extensions. There is no username and password involved as there will be many users coming from this one IP. This is what I have tried recently: [sipin_test] type=peer defaultip=195.27.242.120 context=test_trunk deny=0.0.0.0/0.0.0.0 permit=195.27.242.120/255.255.255.255 dtmfmode=rfc2833 disallow=all allow=ulaw nat=no I have also tried changing what is inside the brackets to the IP address. I have tried many many different combinations of the above, but the IP address never seems to get picked up correctly. I am testing the SIP connection using sipsak. I realize that Asterisk is probably not the best SIP server to use, and plan on migration to SER, but if anyone can offer any suggestions I would really appreciate it. Regards to all, Joe ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Incoming SIP getting in, but not ringing.
Hi all. Just as a quote note, can I thank everyone on this list. I find my self finding pretty much every answer I am looking for on here. And a big thanks to all thoughs helping us out. Mass Respect :) Ok, I'm using a SIP provider (SipGate UK) to do my international dialing etc, working great from extension 8 on phones. However some more friends/contacts have started using SipGate also, and I want to be able to do some SipGate to SipGate calls. As I said I can dial out on SipGate no issues, but I cannot get my [EMAIL PROTECTED] box to receive SipGate calls. I have attached a text file with the sip debug option for a full log. requests are coming in from SipGates server etc but my asterisk box is not transfering the calls to the phones. I have the register string in my sip.conf as so: register=6698221:(MYSECRET)@sipgate.co.uk/6698221 Port on my IPCOP box as follows: UDP/5060 UDP/1:2 UDP/8000:8012 UDP-TCP/3478 Thanks for your time. Paul. Sip read: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Record-Route: sip:[EMAIL PROTECTED];ftag=as6a04ebdf;lr=on Max-Forwards: 9 Record-Route: sip:[EMAIL PROTECTED];ftag=as6a04ebdf;lr=on Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKeafc.0df1fb45.0 Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKeafc.dd5c7802.0 Via: SIP/2.0/UDP 217.10.79.218:5060;branch=z9hG4bK66be0c1c From: 07976xx sip:[EMAIL PROTECTED];tag=as6a04ebdf To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: sipgate asterisk Date: Mon, 10 Oct 2005 15:53:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 448 v=0 o=root 5903 5903 IN IP4 217.10.79.218 s=session c=IN IP4 217.10.79.55 t=0 0 m=audio 44214 RTP/AVP 8 0 3 97 18 2 4 5 110 7 10 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:110 speex/8000 a=rtpmap:7 LPC/8000 a=rtpmap:10 L16/8000 a=silenceSupp:off - - - - a=direction:active a=nortpproxy:yes 17 headers, 20 lines Using latest request as basis request Sending to 217.10.79.219 : 5060 (non-NAT) Found peer 'SipGate' Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKeafc.0df1fb45.0 Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKeafc.dd5c7802.0 Via: SIP/2.0/UDP 217.10.79.218:5060;branch=z9hG4bK66be0c1c From: 07976xx sip:[EMAIL PROTECTED];tag=as6a04ebdf To: sip:[EMAIL PROTECTED];tag=as60d08779 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=557d3579 Content-Length: 0 to 217.10.79.219:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms asterisk1*CLI Sip read: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKeafc.0df1fb45.0 From: 07976xx sip:[EMAIL PROTECTED];tag=as6a04ebdf Call-ID: [EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as60d08779 CSeq: 102 ACK User-Agent: sipgate ser Content-Length: 0 8 headers, 0 lines asterisk1*CLI Sip read: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Record-Route: sip:[EMAIL PROTECTED];ftag=as6a04ebdf;lr=on Max-Forwards: 9 Record-Route: sip:[EMAIL PROTECTED];ftag=as6a04ebdf;lr=on Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKfafc.4aae3986.0 Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKfafc.a85e7d75.0 Via: SIP/2.0/UDP 217.10.79.218:5060;branch=z9hG4bK49e9a0ad From: 07976xx sip:[EMAIL PROTECTED];tag=as6a04ebdf To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: sipgate asterisk Date: Mon, 10 Oct 2005 15:53:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 448 v=0 o=root 5903 5904 IN IP4 217.10.79.218 s=session c=IN IP4 217.10.79.55 t=0 0 m=audio 44214 RTP/AVP 8 0 3 97 18 2 4 5 110 7 10 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:110 speex/8000 a=rtpmap:7 LPC/8000 a=rtpmap:10 L16/8000 a=silenceSupp:off - - - - a=direction:active a=nortpproxy:yes 17 headers, 20 lines Using latest request as basis request Sending to 217.10.79.219 : 5060 (non-NAT) Found peer 'SipGate' Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKfafc.4aae3986.0 Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKfafc.a85e7d75.0 Via: SIP/2.0/UDP 217.10.79.218:5060;branch=z9hG4bK49e9a0ad From: 07976xx sip:[EMAIL PROTECTED];tag=as6a04ebdf To: sip:[EMAIL PROTECTED];tag=as60d08779 Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=70112d01
[Asterisk-Users] Incoming sip
Hi! I use AAH and have 2 sip peers. First one is working perfect both ways. Now I have set up another on and it works perfect for calling out but I get busy when I try to call in. If I use an IP-phone connected directly to the provider it is no problem. Anything special to think about when you have more then 1 provider? Regards Anders Svensson ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] Incoming SIP from Cisco 7206
Here is my entry in sip.conf that works for 7200's, 3600's, and 2600's. [gateway]type=friendhost=192.168.1.61canreinvite=yescontext=gw-inboundqualify=nodtmfmode=rfc2833insecure=yesdisallow=allallow=ulawallow=alaw Hope that helps. B. J. From: Scott Miller [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 03, 2005 16:09To: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] Incoming SIP from Cisco 7206 I am running an Asterisk server through a Cisco 7206 PSTN gateway. I am able to make outgoing SIP calls without a problem, though incoming calls have been somewhat of a problem I am not sure exactly how sip.conf should look in such a scenario. I believe most Cisco gateways are just managed through ACLs, with no authentication, so I think I have the outgoing peer statement right, but I have no idea where to start on the incoming user statement. Heres my sip.conf (configured through AMP) [gk02-inbound] type=user host=10.0.106.10 context=from-pstn [gk01] type=peer host=10.0.50.10 When a call comes it, about every second I get this. Aug 1 11:53:49 DEBUG[4076]: Stopping retransmission on '[EMAIL PROTECTED]' of Response 101: Found Aug 1 11:53:49 DEBUG[4076]: Check for res for Aug 1 11:53:49 DEBUG[4076]: is not a local user Aug 1 11:53:49 DEBUG[4076]: is not a local user Any help would be appreciated. Thanks, Scott Allen Miller Research Assistant Telecommunications University Information Technology Services Indiana University ___ 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] Incoming SIP from Cisco 7206
I am running an Asterisk server through a Cisco 7206 PSTN gateway. I am able to make outgoing SIP calls without a problem, though incoming calls have been somewhat of a problem I am not sure exactly how sip.conf should look in such a scenario. I believe most Cisco gateways are just managed through ACLs, with no authentication, so I think I have the outgoing peer statement right, but I have no idea where to start on the incoming user statement. Heres my sip.conf (configured through AMP) [gk02-inbound] type=user host=10.0.106.10 context=from-pstn [gk01] type=peer host=10.0.50.10 When a call comes it, about every second I get this. Aug 1 11:53:49 DEBUG[4076]: Stopping retransmission on '[EMAIL PROTECTED]' of Response 101: Found Aug 1 11:53:49 DEBUG[4076]: Check for res for Aug 1 11:53:49 DEBUG[4076]: is not a local user Aug 1 11:53:49 DEBUG[4076]: is not a local user Any help would be appreciated. Thanks, Scott Allen Miller Research Assistant Telecommunications University Information Technology Services Indiana University ___ 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] Incoming SIP calls with no extension
Hi All!
I am new to asterisk and have a simple question:
I was able to install and configure it as I wanted. But when I try to
configure a default extension ('s' extension) for incoming sip calls it
doesn't work. I just want that when someone calls my ip it get's
connected to some default extension.
The reason I ask is that I plan to replace a conventional pbx with
asterisk. The setup should include a BRI interface from the telekom
provider. I need to use the same numbers as now because they are well
known. At the moment I just installed asterisk on my laptop to play
with the configuration, and I am afraid that a caller has to use some
extension to call in from the BRI interface when I replace the pbx, just
like now with the software sip phone.
Thanks,
Christoph
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Re: [Asterisk-Users] Incoming SIP calls with no extension
Hello Christoph,
On Sat, 4 Jun 2005, Christoph Weber wrote:
Hi All!
I am new to asterisk and have a simple question:
I was able to install and configure it as I wanted. But when I try to
configure a default extension ('s' extension) for incoming sip calls it
doesn't work. I just want that when someone calls my ip it get's
connected to some default extension.
The reason I ask is that I plan to replace a conventional pbx with
asterisk. The setup should include a BRI interface from the telekom
provider. I need to use the same numbers as now because they are well
known. At the moment I just installed asterisk on my laptop to play
with the configuration, and I am afraid that a caller has to use some
extension to call in from the BRI interface when I replace the pbx, just
like now with the software sip phone.
Set
immediate=no
overlapdial=yes
in zapata.conf
Regards
Torsten
Thanks,
Christoph
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--
Media Online Internet Services Marketing GmbH
Torsten Krueger [EMAIL PROTECTED]
fon: 49-231-5575100fax: 49-231-55751098
Kurze Str. 10 D-44137 Dortmund
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[Asterisk-Users] Incoming SIP calls with different signaling and RTP IP addresses
Hello, I use Asterisk CVS-v1-0-12/21/04-11:05:29 and I noticed that when we receive calls from a partner's IP address (who has a static host entry in the sip.conf file) but the RTP comes from a different address than the signaling, our * sends a 403 forbidden message and drops the call. This problem does not llow us to receive calls from SIP proxies. Was this fixed in newer versions of Asterisk? Best regards, Vlasis. ___ 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
Re: [Asterisk-Users] Incoming SIP Address?
Hi!
[default]
exten = ian,1,Dial(SIP/spa3k_line1,10)
exten = ian,2,Voicemail(u4)
exten = ian,3,Hangup
Is there any way to get such calls coming into a dedicated context,
rather than default?
Use gotoif() and the variable ${SIPDOMAIN}
Cheers, Philipp
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[Asterisk-Users] Incoming SIP Address?
Hi, Is it possible to have an incoming SIP address like [EMAIL PROTECTED], where sip.mydomain.com points to a box running Asterisk? If so, please could someone give an example asterisk config snippet for this? If it is possible, I assume ports 5060 and 1-2 need to be opened in the firewall too. Thanks! --ian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming SIP Address?
Ian Chilton wrote: I assume ports 5060 and 1-2 need to be opened in the firewall too. I don't know much about SIP and firewalls, but opening ten thousand ports doesn't sound good, you've just knocked 1/6 of your firewall down :-( ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Incoming SIP Address?
Hi, Is it possible to have an incoming SIP address like [EMAIL PROTECTED], where sip.mydomain.com points to a box running Asterisk? If so, please could someone give an example asterisk config snippet for this? If it is possible, I assume ports 5060 and 1-2 need to be opened in the firewall too. Thanks! --ian Ian, you don't even have to create a subdomain for this. Include a 'SRV' entry in your DNS record and you can have [EMAIL PROTECTED] http://www.voip-info.org/wiki-DNS+SRV Cheers Shane ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming SIP Address?
Hi, I assume ports 5060 and 1-2 need to be opened in the firewall too. I don't know much about SIP and firewalls, but opening ten thousand ports doesn't sound good, you've just knocked 1/6 of your firewall down That's what I thought but I was told it was the only way to get incoming SIP working when Asterisk was behind a firewall/NAT. I was told it was not a security risk to do this. Any thoughts anyone? --ian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Incoming SIP Address?
Hi, Is it possible to have an incoming SIP address like [EMAIL PROTECTED], where sip.mydomain.com points to a box running Asterisk? If so, please could someone give an example asterisk config snippet for this? snip --ian Ian, you don't even have to create a subdomain for this. Include a 'SRV' entry in your DNS record and you can have [EMAIL PROTECTED] http://www.voip-info.org/wiki-DNS+SRV Cheers Shane Another good link Ian with working examples... http://slacker.com/~nugget/asterisk7.php -Shane ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming SIP Address?
Hi Shane, http://www.voip-info.org/wiki-DNS+SRV http://slacker.com/~nugget/asterisk7.php The SRV page was useful - i've done that in my domain now. But, the other page is talking more about dialing sip addresses through Asterisk rather than incoming sip addresses. However, after adding the SRV record into DNS and the following into Asterisk in extensions.conf, it seems to work: [default] exten = ian,1,Dial(SIP/spa3k_line1,10) exten = ian,2,Voicemail(u4) exten = ian,3,Hangup Is this the right/best way to do it? Is there any way to get such calls coming into a dedicated context, rather than default? Thanks! --ian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming SIP Address?
I assume ports 5060 and 1-2 need to be opened in the firewall too. I don't know much about SIP and firewalls, but opening ten thousand ports doesn't sound good, you've just knocked 1/6 of your firewall down That's what I thought but I was told it was the only way to get incoming SIP working when Asterisk was behind a firewall/NAT. I was told it was not a security risk to do this. Any thoughts anyone? If your configuration and firewall actually require you to open a group of ports to *, then take a look at limiting the rtp ports that are actually used. Examples: - in /etc/asterisk/rtp.conf, look at changing rtpstart and rtpend - for cisco 7960's, look in SIPDefault.cnf for start_media_port and end_media_port - other sip phones often times use other rtp ports, some of which are configurable (and some phones not). Each sip phone vendor use a different range of rtp ports. To reduce the security exposures, one can also use firewall filters to allow only certain external IP addresses (if your firewall supports that function), and/or sip.conf definitions that include something like: deny=0.0.0.0/0.0.0.0 permit=47.136.1.129/255.255.255.0 If you really need to do this, you will almost always need a packet sniffer to see what is actually happening on the inside edge of your firewall and on the outside edge. Without such packet traces changing parameters is nothing more then a guessing game. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming SIP Address?
Ian Chilton wrote: That's what I thought but I was told it was the only way to get incoming SIP working when Asterisk was behind a firewall/NAT. I was told it was not a security risk to do this. If you *know* that only asterisk is listening on the relevant ports it's less of a risk, but it's such a wide range and (in theory at least) leaves plenty of scope for a trojan to listen on one of those ports. Perhaps SElinux can help here, does it allpw you to say that only a cerain process has access to the those ports? Arrghh, I hate the way to:, from: and reply-to: addresses get mangled by lists! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming SIP Address?
Hi Rick, If your configuration and firewall actually require you to open a group of ports to *, then take a look at limiting the rtp ports that are actually used. How many do I need (or how do I find out?) and why does Asterisk specify so many by default? Thanks --ian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming SIP calls not being sent to s extension
I was troubleshooting a problem with incoming calls to my VoicePulse Open Access (NOT Connect) numbers not coming in and I noticed the following in the SIP debug... Found peer 'roamer1-vpoa' Looking for s00** in ivr-incoming Why are the calls getting sent to this weird s00** extension and not the usual s extension in context ivr-incoming as they should? Of course, that s00 extension happens to be the username for my VP Open Access account; I'm thinking that the first letter being an s is confusing Asterisk, since I'm not seeing the same thing with FWD or other services where the username is all numbers. relevant parts of sip.conf for the VP OA account: register = s00**:[EMAIL PROTECTED] ; [roamer1-vpoa] type=friend context=ivr-incoming username=s00** secret=SeCrEt host=access1.voicepulse.com dtmf=inband nat=yes qualify=yes canreinvite=no insecure=very My FWD, SIPPhone, etc. accounts are configured *exactly* the same except for different usernames, secrets, and hosts and they work fine...calls to those numbers go to the s extension in context ivr-incoming as they should. I did come up with a workaround (make a special context contaning the s00 username as an extension that just Gotos the s extension in ivr-incoming), but I shouldn't have to do that... -SC -- Stanley Cline -- sc1 at roamer1 dot org -- http://www.roamer1.org/ ... Never put off until tomorrow what you can do today. There might be a law against it by that time. -/usr/games/fortune ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming SIP gateway context?
I just started service with Broadvoice.com and everything seems to work. However, apparently my understanding of incoming sip contexts is less then what I thought it was. Could someone point me in the right direction? (* on a public address, CVS-HEAD-07/12/04, C7960 phones) In my sip.conf I have: [general] port = 5060 bindaddr = 0.0.0.0 allow=ulaw tos=0x18 ;sets ip tos bits (=lowdelay and throughput) context = bogon-calls ; Send SIP callers that we don't know about here context=from-broadvoice register=303539:[EMAIL PROTECTED]/539 snip [broadvoice] ;this is referenced for outgoing calls to Broadvoice.com type=peer username=303539 snip The problem I'm having with understanding this is for incoming calls from broadvoice. If I remove the context=from-broadvoice from the above, incoming calls from broadvoice are dropped into the bogon-calls context (no service available message). I've tried several different approaches to define another context with type=user, but can never get test calls from broadvoice to be handled in anything other then the bogon-calls context. What am I missing? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming SIP gateway context?
The problem I'm having with understanding this is for incoming calls from broadvoice. If I remove the context=from-broadvoice from the above, incoming calls from broadvoice are dropped into the bogon-calls context (no service available message). just add the context = from-broadvoice to the [broadvoice] section like this: [general] ... context = bogon-calls ; Send SIP callers that we don't know about here ... [broadvoice] type=friend username=303539 host=... context=from-broadvoice i also have a fromuser (value equals username), fromdomain (value equals host) and insecure=very entry in that section to direct incoming calls from sipgate to the right context. as there is no way (other than the originating host) to identify such calls the we context used there should be quite limited. I've tried several different approaches to define another context with type=user, but can never get test calls from broadvoice to be handled in anything other then the bogon-calls context. i use type=friend to handle incoming and outgoing connections in the same section, but you can also define one with type=peer and one with type=user. hope that helps, stefan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming SIP gateway context?
On Sun, 25 Jul 2004, Rich Adamson wrote: I just started service with Broadvoice.com and everything seems to work. However, apparently my understanding of incoming sip contexts is less then what I thought it was. Could someone point me in the right direction? (* on a public address, CVS-HEAD-07/12/04, C7960 phones) In my sip.conf I have: [general] port = 5060 bindaddr = 0.0.0.0 allow=ulaw tos=0x18 ;sets ip tos bits (=lowdelay and throughput) context = bogon-calls ; Send SIP callers that we don't know about here context=from-broadvoice register=303539:[EMAIL PROTECTED]/539 snip [broadvoice] ;this is referenced for outgoing calls to Broadvoice.com type=peer username=303539 snip this doesn't address your question (I think the other post did) but it anticipates your next question.. Add dtmfmode=general to BOTH the general and broadvoice contexts in sip.conf. Asterisk seems to make an incorrect assumption about dtmf with broadvoice (on calls inbound to your box, that is) unless you set it in the general section as well. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming SIP calls as asterisk@...
Hi all, I noticed that all incoming calls come from the user [EMAIL PROTECTED], so I just can't hit the Call button on my SJphone for Linux to return the call... Is there any way to configure Asterisk to show the real [EMAIL PROTECTED] ? Thanks and regards, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] incoming SIP calls drop on pickup.
I also thought it might be a coded mismatch. Maybe someone can explain why outgoing calls work when incoming calls between the same phones don't work? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, March 29, 2004 10:32 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] incoming SIP calls drop on pickup. Sounds like a codec mismatch to me. I had a similar problem with ICH. On Mon, 29 Mar 2004 19:23:15 +0100, jc wrote: Hi All, I have an annoying problem. Out going SIP/sipphone.com calls work fine. Internal calls work fine. However, incoming SIP calls DIAL and ring, but send a busy signal when picked up. The same happens if I take the SNOM200 out of the loop and just try to answer and playback a recording. The debugs don't produce an obvious error. Thanks JC ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] incoming SIP calls drop on pickup.
Hi All, I have an annoying problem. Out going SIP/sipphone.com calls work fine. Internal calls work fine. However, incoming SIP calls DIAL and ring, but send a busy signal when picked up. The same happens if I take the SNOM200 out of the loop and just try to answer and playback a recording. The debugs dont produce an obvious error. Thanks JC
Re: [Asterisk-Users] incoming SIP calls drop on pickup.
Sounds like a codec mismatch to me. I had a similar problem with ICH. On Mon, 29 Mar 2004 19:23:15 +0100, jc wrote: Hi All, I have an annoying problem. Out going SIP/sipphone.com calls work fine. Internal calls work fine. However, incoming SIP calls DIAL and ring, but send a busy signal when picked up. The same happens if I take the SNOM200 out of the loop and just try to answer and playback a recording. The debugs dont produce an obvious error. Thanks JC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming SIP calls
Hello All I am trying to answer incoming SIP calls, first, by dialing an extension, thence into voicemail, which works; and secondly by going straight into voice mail which does not. The extension.conf that works is like this; [incomingSIP] exten=_.,1,Dial,Zap/2|1 exten=_.,2,Voicemail,u5152 exten=_.,3,Hangup the extension.conf which does not is like this; [incomingSIP] exten=_.,1,Answer exten=_.,2,Voicemail,u5152 exten=_.,3,Hangup For the non-working config I cam see the commands being run on the console but the SIP session times out without receiving any audio. I have traced both sessions with ethereal and the protocol handshake is identical however * appears to be ignoring the ACK response for the second config and repeatedly sends 200/OK and then times out. Isuppose I am missing something obvious here but am going 'glassy eyed' trying to spot it. Any help appreciated. Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming SIP-calls and Festival
Hi! I have problems with calls that are coming from a SIP-provider, and where I want to use Festival to play som text to the caller. I hear the text if I call from a SIP-extension (I've tried with g.711a/u and GSM and all three works) But if I call in to the server through my SIP-provider I wont hear any Festival-speech (no error output on the console - see in the end of the mail), if I instead use Background for example I can hear the soundfile. I think it's very strange - is there anyone that have an idea why I can't use Festival with the calls coming from my SIP-provider. This is how it looks on the console - but the caller don't hear anything; --SNIP-- -- Executing Answer(SIP/11292-594f, ) in new stack -- Executing Festival(SIP/11292-594f, 'Hello') in new stack == Parsing '/etc/asterisk/festival.conf': Found == Spawn extension (digisip, 301, 2) exited non-zero on 'SIP/11292-594f' --SNAP-- Regards, Lars --- Lars Fredriksson Ockelbo, Sweden mailto:[EMAIL PROTECTED] http://www.fredriksson.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming SIP matching
I ran some tests and reviewed the source code. It appears that for incoming INVITE messages, Asterisk first checks for [name] entries that match the user portion of the SIP URI in the From: header of the INVITE message.. i.e. if you are calling From sip:[EMAIL PROTECTED] it looks for [123] in the sip.conf file. If this fails then it checks for an IP match. If the IP match fails then it looks in the extensions.conf file (in the context set as default in sip.conf) for a matching extension. If I've intereperted it correctly, it seems a strange way for it to operate. Adding some debug log messages about which sip.conf entry is being selected would make figuring out what is happening a lot easier. Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Thomas B. Clark [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, January 25, 2004 3:01 AM Subject: [Asterisk-Users] Incoming SIP matching Incoming FWD calls from other FWD users, iaxtel, or via ipkall, need to have dtmfmode=rfc2833. However, incoming FWD calls from the dialup access numbers (such as libretel) need to have dtmfmode=inband. To solve this problem, I created a second FWD account and configured sip.conf as follows, in order to match the incoming number to the proper dtmfmode: [fwd-rfc] type=friend secret=* host=dynamic dtmfmode=rfc2833 username=76153 callerid=CLARK THOMAS B 76153 [fwd-inband] type=friend secret=* host=dynamic dtmfmode=inband username=244006 callerid=CLARK THOMAS B 244006 What I find is that, no matter what I change (for example, host-dynamic in order to prevent matching by IP address), I cannot make the incoming SIP calls match successfully. With the configuration above, all incoming calls use dtmfmode=rfc2833, but that could be because it's the default. Either entry works correctly alone (with the other commented out.) I found some discussion in the archives about incoming sip matching, but no patches. Is there a better way to handle the two types of incoming FWD calls? If not, is there something else I could change in order to make them match the correct section? Any ideas would be appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming SIP matching
Incoming FWD calls from other FWD users, iaxtel, or via ipkall, need to have dtmfmode=rfc2833. However, incoming FWD calls from the dialup access numbers (such as libretel) need to have dtmfmode=inband. To solve this problem, I created a second FWD account and configured sip.conf as follows, in order to match the incoming number to the proper dtmfmode: [fwd-rfc] type=friend secret=* host=dynamic dtmfmode=rfc2833 username=76153 callerid=CLARK THOMAS B 76153 [fwd-inband] type=friend secret=* host=dynamic dtmfmode=inband username=244006 callerid=CLARK THOMAS B 244006 What I find is that, no matter what I change (for example, host-dynamic in order to prevent matching by IP address), I cannot make the incoming SIP calls match successfully. With the configuration above, all incoming calls use dtmfmode=rfc2833, but that could be because it's the default. Either entry works correctly alone (with the other commented out.) I found some discussion in the archives about incoming sip matching, but no patches. Is there a better way to handle the two types of incoming FWD calls? If not, is there something else I could change in order to make them match the correct section? Any ideas would be appreciated. Here's hint #1: voipfu*CLI show application SIPDtmfMode -= Info about application 'SIPDtmfMode' =- [Synopsis]: Change the dtmfmode for a SIP call [Description]: SIPDtmfMode(inband|info|rfc2833): Changes the dtmfmode for a SIP call voipfu*CLI Here's hint #2: register=76153:[EMAIL PROTECTED]/76153 Here's hint #3: exten = 76153,1,SIPDtmfMode(rfc2833) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming SIP matching
Incoming FWD calls from other FWD users, iaxtel, or via ipkall, need to have dtmfmode=rfc2833. However, incoming FWD calls from the dialup access numbers (such as libretel) need to have dtmfmode=inband. To solve this problem, I created a second FWD account and configured sip.conf as follows, in order to match the incoming number to the proper dtmfmode: [fwd-rfc] type=friend secret=* host=dynamic dtmfmode=rfc2833 username=76153 callerid=CLARK THOMAS B 76153 [fwd-inband] type=friend secret=* host=dynamic dtmfmode=inband username=244006 callerid=CLARK THOMAS B 244006 What I find is that, no matter what I change (for example, host-dynamic in order to prevent matching by IP address), I cannot make the incoming SIP calls match successfully. With the configuration above, all incoming calls use dtmfmode=rfc2833, but that could be because it's the default. Either entry works correctly alone (with the other commented out.) I found some discussion in the archives about incoming sip matching, but no patches. Is there a better way to handle the two types of incoming FWD calls? If not, is there something else I could change in order to make them match the correct section? Any ideas would be appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
