ITSP inbound ok but cannot complete outbound call
For this ITSP, registration is ok and inbound calls work. But when making an outbound call the call is sent to the itsp but then gets a fast busy. From what I see in the trace the itsp uses asterisk. I sent a trace in a previous email but it shows nothing I could see missing or in error. The itsp sent the config info below but I have been using them without a STUN server fine on my old PBX. Is there anything that looks like it is out of sync with defaults (like port?). Also, the SIP and DP are below but since the call is sent the DP must be ok. Thanks for any help/suggestions!
From the ITSP:> Sip Proxy - sip.voipyourphone.com STN Server - stun.voipyourphone.com Registration Port - any one port between 5060 and 5069 DTMF port 5004 RTP ports 16384 to 16482 Audio Codecs G711u (account default), G711a, G729a and iLbc DTMF rfc2833 (default), inband, SIP-INFO
SIP: <gateway name="uid1"> <param name="proxy" value="sip.voipyourphone.com"/> <param name="username" value="uid1"/> <param name="from-user" value="uid1"/> <param name="password" value="pwd1"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> DP <extension name="local.uid1"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^9(\d{7})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/uid1/1${default_areacode}$1@sip.voipyourlife.com"/> </condition> </extension>
Brand new FS user here. Here is an additional info for my issue of not being able to make outbound calls. I searched and found a "sofia loglevel all 9" command and found that when I make a call I get the 503 service not available message. Inbound works fine. Anyone have an idea? Many thanks! nua(0x8135520): recv signal r_respond 503 Service Unavailable nua: nua_stack_set_params: entering soa_set_params(static::0x812ecc8, ...) called nua: nua_invite_server_respond: entering soa_clear_remote_sdp(static::0x812ecc8) called tport_tsend(0x817aa88) tpn = UDP/10.246.1.27:5060 tport_resolve addrinfo = 10.246.1.27:5060 tport_by_addrinfo(0x817aa88): not found by name UDP/10.246.1.27:5060 tport_vsend(0x817aa88): 838 bytes of 838 to udp/10.246.1.27:5060 tport_vsend returned 838 nta: sent 503 Service Unavailable for INVITE (102) nta: timer shortened to 500 ms nua(0x8135520): removing session usage nua(0x8135520): call state changed: received -> terminated nua(0x8135520): event i_state 503 Service Unavailable nua(0x8135520): event i_terminated 503 Service Unavailable soa_destroy(static::0x812ecc8) called nta_leg_destroy(0x81325f0) nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_handle_bind: entering nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x8135520): sent signal r_destroy nua(0x8135520): recv signal r_destroy nta_leg_destroy((nil)) tport_wakeup_pri(0x817aa88): events IN tport_recv_event(0x817aa88) tport_recv_iovec(0x817aa88) msg 0x81715a8 from (udp/10.246.1.37:5060) has 638 bytes, veclen = 1 tport_deliver(0x817aa88): msg 0x81715a8 (638 bytes) from udp/10.246.1.27:5060/sip next=(nil) nta: received ACK sip:914082558194@10.246.1.37 SIP/2.0 (CSeq 102) nta: ACK (102) is going to INVITE (102) nta: timer set next to 2305 ms nta: timer I fired, terminate 407 response incoming_reclaim_all((nil), (nil), 0xb54550ec) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/3 term, 1/3 free nta: timer set next to 2206 ms nta: timer I fired, terminate 503 response incoming_reclaim_all((nil), (nil), 0xb54550ec) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free On 09/20/10 14:18, Mario wrote:
For this ITSP, registration is ok and inbound calls work. But when making an outbound call the call is sent to the itsp but then gets a fast busy. From what I see in the trace the itsp uses asterisk. I sent a trace in a previous email but it shows nothing I could see missing or in error. The itsp sent the config info below but I have been using them without a STUN server fine on my old PBX. Is there anything that looks like it is out of sync with defaults (like port?). Also, the SIP and DP are below but since the call is sent the DP must be ok. Thanks for any help/suggestions!
From the ITSP:> Sip Proxy - sip.voipyourphone.com STN Server - stun.voipyourphone.com Registration Port - any one port between 5060 and 5069 DTMF port 5004 RTP ports 16384 to 16482 Audio Codecs G711u (account default), G711a, G729a and iLbc DTMF rfc2833 (default), inband, SIP-INFO
SIP: <gateway name="uid1"> <param name="proxy" value="sip.voipyourphone.com"/> <param name="username" value="uid1"/> <param name="from-user" value="uid1"/> <param name="password" value="pwd1"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/>
DP <extension name="local.uid1"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^9(\d{7})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/uid1/1${default_areacode}$1@sip.voipyourlife.com"/> </condition> </extension>
Test the SIP account with a softphone or something. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 20/09/2010 à 23:42, Mario a écrit :
Brand new FS user here. Here is an additional info for my issue of not being able to make outbound calls. I searched and found a "sofia loglevel all 9" command and found that when I make a call I get the 503 service not available message. Inbound works fine. Anyone have an idea? Many thanks!
nua(0x8135520): recv signal r_respond 503 Service Unavailable
nua: nua_stack_set_params: entering soa_set_params(static::0x812ecc8, ...) called nua: nua_invite_server_respond: entering soa_clear_remote_sdp(static::0x812ecc8) called tport_tsend(0x817aa88) tpn = UDP/10.246.1.27:5060 tport_resolve addrinfo = 10.246.1.27:5060 tport_by_addrinfo(0x817aa88): not found by name UDP/10.246.1.27:5060 tport_vsend(0x817aa88): 838 bytes of 838 to udp/10.246.1.27:5060 tport_vsend returned 838 nta: sent 503 Service Unavailable for INVITE (102) nta: timer shortened to 500 ms nua(0x8135520): removing session usage nua(0x8135520): call state changed: received -> terminated nua(0x8135520): event i_state 503 Service Unavailable nua(0x8135520): event i_terminated 503 Service Unavailable soa_destroy(static::0x812ecc8) called nta_leg_destroy(0x81325f0) nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_handle_bind: entering nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x8135520): sent signal r_destroy nua(0x8135520): recv signal r_destroy nta_leg_destroy((nil)) tport_wakeup_pri(0x817aa88): events IN tport_recv_event(0x817aa88) tport_recv_iovec(0x817aa88) msg 0x81715a8 from (udp/10.246.1.37:5060) has 638 bytes, veclen = 1 tport_deliver(0x817aa88): msg 0x81715a8 (638 bytes) from udp/10.246.1.27:5060/sip next=(nil) nta: received ACK sip:914082558194@10.246.1.37 SIP/2.0 (CSeq 102) nta: ACK (102) is going to INVITE (102) nta: timer set next to 2305 ms nta: timer I fired, terminate 407 response incoming_reclaim_all((nil), (nil), 0xb54550ec) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/3 term, 1/3 free nta: timer set next to 2206 ms nta: timer I fired, terminate 503 response incoming_reclaim_all((nil), (nil), 0xb54550ec) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free
On 09/20/10 14:18, Mario wrote:
For this ITSP, registration is ok and inbound calls work. But when making an outbound call the call is sent to the itsp but then gets a fast busy. From what I see in the trace the itsp uses asterisk. I sent a trace in a previous email but it shows nothing I could see missing or in error. The itsp sent the config info below but I have been using them without a STUN server fine on my old PBX. Is there anything that looks like it is out of sync with defaults (like port?). Also, the SIP and DP are below but since the call is sent the DP must be ok. Thanks for any help/suggestions!
From the ITSP:> Sip Proxy - sip.voipyourphone.com STN Server - stun.voipyourphone.com Registration Port - any one port between 5060 and 5069 DTMF port 5004 RTP ports 16384 to 16482 Audio Codecs G711u (account default), G711a, G729a and iLbc DTMF rfc2833 (default), inband, SIP-INFO
SIP: <gateway name="uid1"> <param name="proxy" value="sip.voipyourphone.com"/> <param name="username" value="uid1"/> <param name="from-user" value="uid1"/> <param name="password" value="pwd1"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/>
DP <extension name="local.uid1"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^9(\d{7})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/uid1/1${default_areacode} $1@sip.voipyourlife.com"/> </condition> </extension>
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This ITSP account is working for 2 years on a linksys/CICS PBX, SPA962 phones, and any softphone. It works now. One other ITSP works fine in and out, Only on FS do I have the problem with this ITSP. I have 2 new items after talking to the ITSP: 1. I noticed I am getting 407 Proxy Authentication Required messages when I try to make a call. Inbound ok. 2. The ITSP says he sees no return IP address for the call. I am behind a firewall with NAT and all ports open for testing. The Linksys PBX and other phones work fine. The other ITSP also works fine with FS. It's only this one ITSP. I am trying to find keywords on authentication for the gateway parms if there are any. Mario On 09/20/10 15:31, David Ponzone wrote:
Test the SIP account with a softphone or something.
David Ponzone Direction Technique email: david.ponzone@ipeva.fr <mailto:david.ponzone@ipeva.fr> tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr <BLOCKED::http://www.ipeva.fr/> - www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
/Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. /*/IPeva/*/ décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur./ / /
Le 20/09/2010 à 23:42, Mario a écrit :
Brand new FS user here. Here is an additional info for my issue of not being able to make outbound calls. I searched and found a "sofia loglevel all 9" command and found that when I make a call I get the 503 service not available message. Inbound works fine. Anyone have an idea? Many thanks!
nua(0x8135520): recv signal r_respond 503 Service Unavailable
nua: nua_stack_set_params: entering soa_set_params(static::0x812ecc8, ...) called nua: nua_invite_server_respond: entering soa_clear_remote_sdp(static::0x812ecc8) called tport_tsend(0x817aa88) tpn = UDP/10.246.1.27:5060 tport_resolve addrinfo = 10.246.1.27:5060 tport_by_addrinfo(0x817aa88): not found by name UDP/10.246.1.27:5060 tport_vsend(0x817aa88): 838 bytes of 838 to udp/10.246.1.27:5060 tport_vsend returned 838 nta: sent 503 Service Unavailable for INVITE (102) nta: timer shortened to 500 ms nua(0x8135520): removing session usage nua(0x8135520): call state changed: received -> terminated nua(0x8135520): event i_state 503 Service Unavailable nua(0x8135520): event i_terminated 503 Service Unavailable soa_destroy(static::0x812ecc8) called nta_leg_destroy(0x81325f0) nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_handle_bind: entering nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x8135520): sent signal r_destroy nua(0x8135520): recv signal r_destroy nta_leg_destroy((nil)) tport_wakeup_pri(0x817aa88): events IN tport_recv_event(0x817aa88) tport_recv_iovec(0x817aa88) msg 0x81715a8 from (udp/10.246.1.37:5060) has 638 bytes, veclen = 1 tport_deliver(0x817aa88): msg 0x81715a8 (638 bytes) from udp/10.246.1.27:5060/sip next=(nil) nta: received ACK sip:914082558194@10.246.1.37 SIP/2.0 (CSeq 102) nta: ACK (102) is going to INVITE (102) nta: timer set next to 2305 ms nta: timer I fired, terminate 407 response incoming_reclaim_all((nil), (nil), 0xb54550ec) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/3 term, 1/3 free nta: timer set next to 2206 ms nta: timer I fired, terminate 503 response incoming_reclaim_all((nil), (nil), 0xb54550ec) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free
On 09/20/10 14:18, Mario wrote:
For this ITSP, registration is ok and inbound calls work. But when making an outbound call the call is sent to the itsp but then gets a fast busy. From what I see in the trace the itsp uses asterisk. I sent a trace in a previous email but it shows nothing I could see missing or in error. The itsp sent the config info below but I have been using them without a STUN server fine on my old PBX. Is there anything that looks like it is out of sync with defaults (like port?). Also, the SIP and DP are below but since the call is sent the DP must be ok. Thanks for any help/suggestions!
From the ITSP:> Sip Proxy - sip.voipyourphone.com STN Server - stun.voipyourphone.com Registration Port - any one port between 5060 and 5069 DTMF port 5004 RTP ports 16384 to 16482 Audio Codecs G711u (account default), G711a, G729a and iLbc DTMF rfc2833 (default), inband, SIP-INFO
SIP: <gateway name="uid1"> <param name="proxy" value="sip.voipyourphone.com"/> <param name="username" value="uid1"/> <param name="from-user" value="uid1"/> <param name="password" value="pwd1"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/>
DP <extension name="local.uid1"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^9(\d{7})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/uid1/1${default_areacode}$1@sip.voipyourlife.com <mailto:sofia/gateway/uid1/1${default_areacode}$1@sip.voipyourlife.com>"/> </condition> </extension>
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Appreciate anyone who can point me to the right direction. Here is more info from logelevel 9 trace. I ran a test with the working ITSP and non-working ITSP and compared the trace. They are below and it is clear where things are different. Unfortunately, I don't see how to solve the problem. At first I thought it was the "Auth Required" messages but I see them in the ITSP1 trace that works also. So I don't know if that's the problem. Eve if it is I don't see how to fix it since the gateway parms look good. This is my only major problem with FS at this point, I am new to FS but have most things working based on the new book. --------------- ITSP2 that does NOT work ----------------------------- nta: sent 100 Trying for INVITE (101) nua(0xb67056e0): event i_invite 100 Trying nua(0xb67056e0): call state changed: init -> received, received offer soa_get_remote_sdp(static::0xb67048e0, [0xb5373cac], [0xb5373ca8], [(nil)]) called nua(0xb67056e0): event i_state 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0xb67056e0): sent signal r_respond nua: nua_handle_magic: entering nua: nua_handle_destroy: entering <------- DIFFERENCE BELOW nua(0xb67056e0): sent signal r_destroy nua: nua_application_event: entering nua(0xb67056e0): event i_state dropped nua(0xb67056e0): recv signal r_respond 407 Proxy Authentication Required nua: nua_stack_set_params: entering soa_set_params(static::0xb67048e0, ...) called nua: nua_invite_server_respond: entering soa_clear_remote_sdp(static::0xb67048e0) called tport_tsend(0x81c1870) tpn = UDP/10.246.1.27:5060 tport_resolve addrinfo = 10.246.1.27:5060 tport_by_addrinfo(0x81c1870): not found by name UDP/10.246.1.27:5060 tport_vsend(0x81c1870): 813 bytes of 813 to udp/10.246.1.27:5060 tport_vsend returned 813 nta: sent 407 Proxy Authentication Required for INVITE (101) --------------- ITSP1 that works ---------------------------------- nta: sent 100 Trying for INVITE (102) nua(0x827cc00): event i_invite 100 Trying nua(0x827cc00): call state changed: init -> received, received offer soa_get_remote_sdp(static::0x827d0b8, [0xb53a9cac], [0xb53a9ca8], [(nil)]) called nua(0x827cc00): event i_state 100 Trying nua: nua_application_event: entering nta: timer set next to 4910 ms 2010-09-20 16:34:40.367812 [NOTICE] switch_channel.c:779 New Channel sofia/internal/200@10.246.1.37 [39ddcdbe-2f0e-4c58-a492-707c04baec1e] nua: nua_handle_bind: entering nua: nua_handle_magic: entering nua: nua_application_event: entering <- DIFFERENCE FROM ABOVE nua: nua_handle_magic: entering 2010-09-20 16:34:40.520174 [INFO] mod_dialplan_xml.c:331 Processing 200 <200>->714082558194 in context default 2010-09-20 16:34:40.531634 [NOTICE] switch_channel.c:779 New Channel sofia/external/14082558194@callcentric.com [8011f48a-3494-4e92-a5c3-eadc7139530d] nua: nh_create_handle: entering nua: nua_handle_bind: entering nua: nua_invite: entering nua(0x83e2a68): sent signal r_invite nua(0x83e2a68): recv signal r_invite nua: nua_stack_set_params: entering soa_clone(static::0x81799a0, 0x82d6048, 0x83e2a68) called soa_set_params(static::0x82081a8, ...) called soa_set_params(static::0x82081a8, ...) called On 09/20/10 15:47, Mario wrote:
This ITSP account is working for 2 years on a linksys/CICS PBX, SPA962 phones, and any softphone. It works now. One other ITSP works fine in and out, Only on FS do I have the problem with this ITSP. I have 2 new items after talking to the ITSP:
1. I noticed I am getting 407 Proxy Authentication Required messages when I try to make a call. Inbound ok.
2. The ITSP says he sees no return IP address for the call. I am behind a firewall with NAT and all ports open for testing. The Linksys PBX and other phones work fine. The other ITSP also works fine with FS. It's only this one ITSP. I am trying to find keywords on authentication for the gateway parms if there are any. Mario
On 09/20/10 15:31, David Ponzone wrote:
Test the SIP account with a softphone or something.
David Ponzone Direction Technique email: david.ponzone@ipeva.fr <mailto:david.ponzone@ipeva.fr> tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr <BLOCKED::http://www.ipeva.fr/> - www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
/Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. /*/IPeva/*/ décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur./ / /
Le 20/09/2010 à 23:42, Mario a écrit :
Brand new FS user here. Here is an additional info for my issue of not being able to make outbound calls. I searched and found a "sofia loglevel all 9" command and found that when I make a call I get the 503 service not available message. Inbound works fine. Anyone have an idea? Many thanks!
nua(0x8135520): recv signal r_respond 503 Service Unavailable
nua: nua_stack_set_params: entering soa_set_params(static::0x812ecc8, ...) called nua: nua_invite_server_respond: entering soa_clear_remote_sdp(static::0x812ecc8) called tport_tsend(0x817aa88) tpn = UDP/10.246.1.27:5060 tport_resolve addrinfo = 10.246.1.27:5060 tport_by_addrinfo(0x817aa88): not found by name UDP/10.246.1.27:5060 tport_vsend(0x817aa88): 838 bytes of 838 to udp/10.246.1.27:5060 tport_vsend returned 838 nta: sent 503 Service Unavailable for INVITE (102) nta: timer shortened to 500 ms nua(0x8135520): removing session usage nua(0x8135520): call state changed: received -> terminated nua(0x8135520): event i_state 503 Service Unavailable nua(0x8135520): event i_terminated 503 Service Unavailable soa_destroy(static::0x812ecc8) called nta_leg_destroy(0x81325f0) nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_handle_bind: entering nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x8135520): sent signal r_destroy nua(0x8135520): recv signal r_destroy nta_leg_destroy((nil)) tport_wakeup_pri(0x817aa88): events IN tport_recv_event(0x817aa88) tport_recv_iovec(0x817aa88) msg 0x81715a8 from (udp/10.246.1.37:5060) has 638 bytes, veclen = 1 tport_deliver(0x817aa88): msg 0x81715a8 (638 bytes) from udp/10.246.1.27:5060/sip next=(nil) nta: received ACK sip:914082558194@10.246.1.37 SIP/2.0 (CSeq 102) nta: ACK (102) is going to INVITE (102) nta: timer set next to 2305 ms nta: timer I fired, terminate 407 response incoming_reclaim_all((nil), (nil), 0xb54550ec) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/3 term, 1/3 free nta: timer set next to 2206 ms nta: timer I fired, terminate 503 response incoming_reclaim_all((nil), (nil), 0xb54550ec) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free
On 09/20/10 14:18, Mario wrote:
For this ITSP, registration is ok and inbound calls work. But when making an outbound call the call is sent to the itsp but then gets a fast busy. From what I see in the trace the itsp uses asterisk. I sent a trace in a previous email but it shows nothing I could see missing or in error. The itsp sent the config info below but I have been using them without a STUN server fine on my old PBX. Is there anything that looks like it is out of sync with defaults (like port?). Also, the SIP and DP are below but since the call is sent the DP must be ok. Thanks for any help/suggestions!
From the ITSP:> Sip Proxy - sip.voipyourphone.com STN Server - stun.voipyourphone.com Registration Port - any one port between 5060 and 5069 DTMF port 5004 RTP ports 16384 to 16482 Audio Codecs G711u (account default), G711a, G729a and iLbc DTMF rfc2833 (default), inband, SIP-INFO
SIP: <gateway name="uid1"> <param name="proxy" value="sip.voipyourphone.com"/> <param name="username" value="uid1"/> <param name="from-user" value="uid1"/> <param name="password" value="pwd1"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/>
DP <extension name="local.uid1"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^9(\d{7})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/uid1/1${default_areacode}$1@sip.voipyourlife.com <mailto:sofia/gateway/uid1/1${default_areacode}$1@sip.voipyourlife.com>"/> </condition> </extension>
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Can you send us the output of: sofia status profile <profile> on all the profiles you used to register to your ITSPs David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 21/09/2010 à 01:58, Mario a écrit :
Appreciate anyone who can point me to the right direction. Here is more info from logelevel 9 trace. I ran a test with the working ITSP and non-working ITSP and compared the trace. They are below and it is clear where things are different. Unfortunately, I don't see how to solve the problem. At first I thought it was the "Auth Required" messages but I see them in the ITSP1 trace that works also. So I don't know if that's the problem. Eve if it is I don't see how to fix it since the gateway parms look good. This is my only major problem with FS at this point, I am new to FS but have most things working based on the new book.
--------------- ITSP2 that does NOT work ----------------------------- nta: sent 100 Trying for INVITE (101) nua(0xb67056e0): event i_invite 100 Trying nua(0xb67056e0): call state changed: init -> received, received offer soa_get_remote_sdp(static::0xb67048e0, [0xb5373cac], [0xb5373ca8], [(nil)]) called nua(0xb67056e0): event i_state 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0xb67056e0): sent signal r_respond nua: nua_handle_magic: entering nua: nua_handle_destroy: entering <------- DIFFERENCE BELOW nua(0xb67056e0): sent signal r_destroy nua: nua_application_event: entering nua(0xb67056e0): event i_state dropped nua(0xb67056e0): recv signal r_respond 407 Proxy Authentication Required nua: nua_stack_set_params: entering soa_set_params(static::0xb67048e0, ...) called nua: nua_invite_server_respond: entering soa_clear_remote_sdp(static::0xb67048e0) called tport_tsend(0x81c1870) tpn = UDP/10.246.1.27:5060 tport_resolve addrinfo = 10.246.1.27:5060 tport_by_addrinfo(0x81c1870): not found by name UDP/10.246.1.27:5060 tport_vsend(0x81c1870): 813 bytes of 813 to udp/10.246.1.27:5060 tport_vsend returned 813 nta: sent 407 Proxy Authentication Required for INVITE (101)
--------------- ITSP1 that works ---------------------------------- nta: sent 100 Trying for INVITE (102) nua(0x827cc00): event i_invite 100 Trying nua(0x827cc00): call state changed: init -> received, received offer soa_get_remote_sdp(static::0x827d0b8, [0xb53a9cac], [0xb53a9ca8], [(nil)]) called nua(0x827cc00): event i_state 100 Trying nua: nua_application_event: entering nta: timer set next to 4910 ms 2010-09-20 16:34:40.367812 [NOTICE] switch_channel.c:779 New Channel sofia/internal/200@10.246.1.37 [39ddcdbe-2f0e-4c58-a492-707c04baec1e] nua: nua_handle_bind: entering
nua: nua_handle_magic: entering nua: nua_application_event: entering <- DIFFERENCE FROM ABOVE nua: nua_handle_magic: entering 2010-09-20 16:34:40.520174 [INFO] mod_dialplan_xml.c:331 Processing 200 <200>->714082558194 in context default 2010-09-20 16:34:40.531634 [NOTICE] switch_channel.c:779 New Channel sofia/external/14082558194@callcentric.com [8011f48a-3494-4e92-a5c3-eadc7139530d] nua: nh_create_handle: entering
nua: nua_handle_bind: entering nua: nua_invite: entering nua(0x83e2a68): sent signal r_invite nua(0x83e2a68): recv signal r_invite nua: nua_stack_set_params: entering soa_clone(static::0x81799a0, 0x82d6048, 0x83e2a68) called soa_set_params(static::0x82081a8, ...) called soa_set_params(static::0x82081a8, ...) called
On 09/20/10 15:47, Mario wrote:
This ITSP account is working for 2 years on a linksys/CICS PBX, SPA962 phones, and any softphone. It works now. One other ITSP works fine in and out, Only on FS do I have the problem with this ITSP. I have 2 new items after talking to the ITSP:
1. I noticed I am getting 407 Proxy Authentication Required messages when I try to make a call. Inbound ok.
2. The ITSP says he sees no return IP address for the call. I am behind a firewall with NAT and all ports open for testing. The Linksys PBX and other phones work fine. The other ITSP also works fine with FS. It's only this one ITSP. I am trying to find keywords on authentication for the gateway parms if there are any. Mario
On 09/20/10 15:31, David Ponzone wrote:
Test the SIP account with a softphone or something.
David Ponzone Direction Technique email: david.ponzone@ipeva.fr <mailto:david.ponzone@ipeva.fr> tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr <BLOCKED::http://www.ipeva.fr/> - www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
/Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. /*/IPeva/*/ décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur./ / /
Le 20/09/2010 à 23:42, Mario a écrit :
Brand new FS user here. Here is an additional info for my issue of not being able to make outbound calls. I searched and found a "sofia loglevel all 9" command and found that when I make a call I get the 503 service not available message. Inbound works fine. Anyone have an idea? Many thanks!
nua(0x8135520): recv signal r_respond 503 Service Unavailable
nua: nua_stack_set_params: entering soa_set_params(static::0x812ecc8, ...) called nua: nua_invite_server_respond: entering soa_clear_remote_sdp(static::0x812ecc8) called tport_tsend(0x817aa88) tpn = UDP/10.246.1.27:5060 tport_resolve addrinfo = 10.246.1.27:5060 tport_by_addrinfo(0x817aa88): not found by name UDP/ 10.246.1.27:5060 tport_vsend(0x817aa88): 838 bytes of 838 to udp/10.246.1.27:5060 tport_vsend returned 838 nta: sent 503 Service Unavailable for INVITE (102) nta: timer shortened to 500 ms nua(0x8135520): removing session usage nua(0x8135520): call state changed: received -> terminated nua(0x8135520): event i_state 503 Service Unavailable nua(0x8135520): event i_terminated 503 Service Unavailable soa_destroy(static::0x812ecc8) called nta_leg_destroy(0x81325f0) nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_handle_bind: entering nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x8135520): sent signal r_destroy nua(0x8135520): recv signal r_destroy nta_leg_destroy((nil)) tport_wakeup_pri(0x817aa88): events IN tport_recv_event(0x817aa88) tport_recv_iovec(0x817aa88) msg 0x81715a8 from (udp/ 10.246.1.37:5060) has 638 bytes, veclen = 1 tport_deliver(0x817aa88): msg 0x81715a8 (638 bytes) from udp/10.246.1.27:5060/sip next=(nil) nta: received ACK sip:914082558194@10.246.1.37 SIP/2.0 (CSeq 102) nta: ACK (102) is going to INVITE (102) nta: timer set next to 2305 ms nta: timer I fired, terminate 407 response incoming_reclaim_all((nil), (nil), 0xb54550ec) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/3 term, 1/3 free nta: timer set next to 2206 ms nta: timer I fired, terminate 503 response incoming_reclaim_all((nil), (nil), 0xb54550ec) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free
On 09/20/10 14:18, Mario wrote:
For this ITSP, registration is ok and inbound calls work. But when making an outbound call the call is sent to the itsp but then gets a fast busy. From what I see in the trace the itsp uses asterisk. I sent a trace in a previous email but it shows nothing I could see missing or in error. The itsp sent the config info below but I have been using them without a STUN server fine on my old PBX. Is there anything that looks like it is out of sync with defaults (like port?). Also, the SIP and DP are below but since the call is sent the DP must be ok. Thanks for any help/suggestions!
From the ITSP:> Sip Proxy - sip.voipyourphone.com STN Server - stun.voipyourphone.com Registration Port - any one port between 5060 and 5069 DTMF port 5004 RTP ports 16384 to 16482 Audio Codecs G711u (account default), G711a, G729a and iLbc DTMF rfc2833 (default), inband, SIP-INFO
SIP: <gateway name="uid1"> <param name="proxy" value="sip.voipyourphone.com"/> <param name="username" value="uid1"/> <param name="from-user" value="uid1"/> <param name="password" value="pwd1"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/>
DP <extension name="local.uid1"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^9(\d{7})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/uid1/1${default_areacode}$1@sip.voipyourlife.com <mailto:sofia/gateway/uid1/1${default_areacode}$1@sip.voipyourlife.com
"/> </condition> </extension>
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I was on the phone with the ITSP for the third time and found someone who looked at their end again. He said it looked like a familiar DNS error he had seen. When I went through the gateway and dialplan names with him (the other techs didn't want to) we noticed that the diaplan had a different name than the gateway and found an error in the asterisk setup info they sent me (and others), the gateway and dialplan are different. You can see the gateway setup down below. WRONG: <action application="bridge" data="sofia/gateway/uid1/1${default_areacode}$1@sip.voipyourlfe.com"/> RIGHT: <action application="bridge" data="sofia/gateway/uid1/1${default_areacode}$1@sip.voipyourphone.com"/> This was a documentation problem that was really tough to find even with the traces. This is also why inbound worked fine and outbound connected to them but the call never completed. Although it was costly for a day of my time I did learn a lot and hope this helps someone else. Thank you all for your generous help! Mario On 09/21/10 01:29, David Ponzone wrote:
Can you send us the output of: sofia status profile <profile> on all the profiles you used to register to your ITSPs
David Ponzone Direction Technique email: david.ponzone@ipeva.fr <mailto:david.ponzone@ipeva.fr> tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr <BLOCKED::http://www.ipeva.fr/> - www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
/Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. /*/IPeva/*/ décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur./ / /
Le 21/09/2010 à 01:58, Mario a écrit :
Appreciate anyone who can point me to the right direction. Here is more info from logelevel 9 trace. I ran a test with the working ITSP and non-working ITSP and compared the trace. They are below and it is clear where things are different. Unfortunately, I don't see how to solve the problem. At first I thought it was the "Auth Required" messages but I see them in the ITSP1 trace that works also. So I don't know if that's the problem. Eve if it is I don't see how to fix it since the gateway parms look good. This is my only major problem with FS at this point, I am new to FS but have most things working based on the new book.
--------------- ITSP2 that does NOT work ----------------------------- nta: sent 100 Trying for INVITE (101) nua(0xb67056e0): event i_invite 100 Trying nua(0xb67056e0): call state changed: init -> received, received offer soa_get_remote_sdp(static::0xb67048e0, [0xb5373cac], [0xb5373ca8], [(nil)]) called nua(0xb67056e0): event i_state 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0xb67056e0): sent signal r_respond nua: nua_handle_magic: entering nua: nua_handle_destroy: entering <------- DIFFERENCE BELOW nua(0xb67056e0): sent signal r_destroy nua: nua_application_event: entering nua(0xb67056e0): event i_state dropped nua(0xb67056e0): recv signal r_respond 407 Proxy Authentication Required nua: nua_stack_set_params: entering soa_set_params(static::0xb67048e0, ...) called nua: nua_invite_server_respond: entering soa_clear_remote_sdp(static::0xb67048e0) called tport_tsend(0x81c1870) tpn = UDP/10.246.1.27:5060 tport_resolve addrinfo = 10.246.1.27:5060 tport_by_addrinfo(0x81c1870): not found by name UDP/10.246.1.27:5060 tport_vsend(0x81c1870): 813 bytes of 813 to udp/10.246.1.27:5060 tport_vsend returned 813 nta: sent 407 Proxy Authentication Required for INVITE (101)
--------------- ITSP1 that works ---------------------------------- nta: sent 100 Trying for INVITE (102) nua(0x827cc00): event i_invite 100 Trying nua(0x827cc00): call state changed: init -> received, received offer soa_get_remote_sdp(static::0x827d0b8, [0xb53a9cac], [0xb53a9ca8], [(nil)]) called nua(0x827cc00): event i_state 100 Trying nua: nua_application_event: entering nta: timer set next to 4910 ms 2010-09-20 16:34:40.367812 [NOTICE] switch_channel.c:779 New Channel sofia/internal/200@10.246.1.37 <mailto:sofia/internal/200@10.246.1.37> [39ddcdbe-2f0e-4c58-a492-707c04baec1e] nua: nua_handle_bind: entering
nua: nua_handle_magic: entering nua: nua_application_event: entering <- DIFFERENCE FROM ABOVE nua: nua_handle_magic: entering 2010-09-20 16:34:40.520174 [INFO] mod_dialplan_xml.c:331 Processing 200 <200>->719161234567 in context default 2010-09-20 16:34:40.531634 [NOTICE] switch_channel.c:779 New Channel sofia/external/19161234567@callcentric.com <mailto:sofia/external/19161234567@callcentric.com> [8011f48a-3494-4e92-a5c3-eadc7139530d] nua: nh_create_handle: entering
nua: nua_handle_bind: entering nua: nua_invite: entering nua(0x83e2a68): sent signal r_invite nua(0x83e2a68): recv signal r_invite nua: nua_stack_set_params: entering soa_clone(static::0x81799a0, 0x82d6048, 0x83e2a68) called soa_set_params(static::0x82081a8, ...) called soa_set_params(static::0x82081a8, ...) called
On 09/20/10 15:47, Mario wrote:
This ITSP account is working for 2 years on a linksys/CICS PBX, SPA962 phones, and any softphone. It works now. One other ITSP works fine in and out, Only on FS do I have the problem with this ITSP. I have 2 new items after talking to the ITSP:
1. I noticed I am getting 407 Proxy Authentication Required messages when I try to make a call. Inbound ok.
2. The ITSP says he sees no return IP address for the call. I am behind a firewall with NAT and all ports open for testing. The Linksys PBX and other phones work fine. The other ITSP also works fine with FS. It's only this one ITSP. I am trying to find keywords on authentication for the gateway parms if there are any. Mario
On 09/20/10 15:31, David Ponzone wrote:
Test the SIP account with a softphone or something.
David Ponzone Direction Technique email: david.ponzone@ipeva.fr <mailto:david.ponzone@ipeva.fr> tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr <BLOCKED::http://www.ipeva.fr/> - www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
/Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. /*/IPeva/*/ décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur./ / /
Le 20/09/2010 à 23:42, Mario a écrit :
Brand new FS user here. Here is an additional info for my issue of not being able to make outbound calls. I searched and found a "sofia loglevel all 9" command and found that when I make a call I get the 503 service not available message. Inbound works fine. Anyone have an idea? Many thanks!
nua(0x8135520): recv signal r_respond 503 Service Unavailable
nua: nua_stack_set_params: entering soa_set_params(static::0x812ecc8, ...) called nua: nua_invite_server_respond: entering soa_clear_remote_sdp(static::0x812ecc8) called tport_tsend(0x817aa88) tpn = UDP/10.246.1.27:5060 tport_resolve addrinfo = 10.246.1.27:5060 tport_by_addrinfo(0x817aa88): not found by name UDP/10.246.1.27:5060 tport_vsend(0x817aa88): 838 bytes of 838 to udp/10.246.1.27:5060 tport_vsend returned 838 nta: sent 503 Service Unavailable for INVITE (102) nta: timer shortened to 500 ms nua(0x8135520): removing session usage nua(0x8135520): call state changed: received -> terminated nua(0x8135520): event i_state 503 Service Unavailable nua(0x8135520): event i_terminated 503 Service Unavailable soa_destroy(static::0x812ecc8) called nta_leg_destroy(0x81325f0) nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_handle_bind: entering nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x8135520): sent signal r_destroy nua(0x8135520): recv signal r_destroy nta_leg_destroy((nil)) tport_wakeup_pri(0x817aa88): events IN tport_recv_event(0x817aa88) tport_recv_iovec(0x817aa88) msg 0x81715a8 from (udp/10.246.1.37:5060) has 638 bytes, veclen = 1 tport_deliver(0x817aa88): msg 0x81715a8 (638 bytes) from udp/10.246.1.27:5060/sip next=(nil) nta: received ACK sip:919161234567@10.246.1.37 SIP/2.0 (CSeq 102) nta: ACK (102) is going to INVITE (102) nta: timer set next to 2305 ms nta: timer I fired, terminate 407 response incoming_reclaim_all((nil), (nil), 0xb54550ec) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/3 term, 1/3 free nta: timer set next to 2206 ms nta: timer I fired, terminate 503 response incoming_reclaim_all((nil), (nil), 0xb54550ec) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free
On 09/20/10 14:18, Mario wrote:
For this ITSP, registration is ok and inbound calls work. But when making an outbound call the call is sent to the itsp but then gets a fast busy. From what I see in the trace the itsp uses asterisk. I sent a trace in a previous email but it shows nothing I could see missing or in error. The itsp sent the config info below but I have been using them without a STUN server fine on my old PBX. Is there anything that looks like it is out of sync with defaults (like port?). Also, the SIP and DP are below but since the call is sent the DP must be ok. Thanks for any help/suggestions!
From the ITSP:> Sip Proxy - sip.voipyourphone.com STN Server - stun.voipyourphone.com Registration Port - any one port between 5060 and 5069 DTMF port 5004 RTP ports 16384 to 16482 Audio Codecs G711u (account default), G711a, G729a and iLbc DTMF rfc2833 (default), inband, SIP-INFO
SIP: <gateway name="uid1"> <param name="proxy" value="sip.voipyourphone.com"/> <param name="username" value="uid1"/> <param name="from-user" value="uid1"/> <param name="password" value="pwd1"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/>
DP <extension name="local.uid1"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^9(\d{7})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/uid1/1${default_areacode}$1@sip.voipyourlife.com <mailto:sofia/gateway/uid1/1${default_areacode}$1@sip.voipyourlife.com>"/> </condition> </extension>
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That is on the FreeSwitch sample on the Callcentric support page. I looked at the new FS book and the wiki for when to supply @hostnmame but could not find an explanation so I used the sample which works. Thanks. Now I know. Who is cooter? On 09/21/2010 08:42 AM, Brian West wrote:
Wrong again cooter! You don't put an @hostname on dialing via providers.
/b
On Sep 21, 2010, at 10:30 AM, Mario wrote:
RIGHT: <action application="bridge" data="sofia/gateway/uid1/1${default_areacode}$1@sip.voipyourphone.com"/>
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-- *Mario*
You don't provide a @hostname when dialing via a gateway. /b On Sep 21, 2010, at 10:57 AM, Mario wrote:
That is on the FreeSwitch sample on the Callcentric support page. I looked at the new FS book and the wiki for when to supply @hostnmame but could not find an explanation so I used the sample which works. Thanks. Now I know. Who is cooter?
On 09/21/2010 08:42 AM, Brian West wrote:
Wrong again cooter! You don't put an @hostname on dialing via providers.
/b
participants (3)
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Brian West
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David Ponzone
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Mario