# Resolving issue **URL:** \ **Category:** Asterisk SIP\ **Created:** [September 29, 2020, 3:58am UTC](https://community.asterisk.org/t/resolving-issue/85861 "2020-09-29T03:58:29Z")\ **Posts on this page:** 8\ **Page:** 1
**Author:** ![sergio](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/sergio/32/4350_2.png) [@sergio](https://community.asterisk.org/u/sergio)\ **Post date:** [September 29, 2020, 3:58am UTC](https://community.asterisk.org/t/resolving-issue/85861/1 "2020-09-29T03:58:29Z")
I have two records in dialplan: ```auto exten => testA,1,Dial(PJSIP/outgoing/sip:thetestcall@sip.linphone.org) exten => testB,1,Dial(PJSIP/outgoing/sip:thetestcall@iptel.org) ``` Calling `testA` works fine, trying to call `testB` gives: ```auto -- Executing [testB@context:1] Dial("PJSIP/client-00000002", "PJSIP/outgoing/sip:thetestcall@iptel.org") in new stack -- Called PJSIP/outgoing/sip:thetestcall@iptel.org == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'PJSIP/client-00000002' status is 'CONGESTION' <--- Transmitting SIP response (342 bytes) to TLS::42435 ---> SIP/2.0 503 Service Unavailable v: SIP/2.0/TLS :42435;rport=42435;received=;branch=z9hG4bK.sdLpOW0sj i: cPXhYEPQLW f: ;tag=6YBHFYM0j t: ;tag=3c93e8ac-908d-4049-bf19-84eec4b967e1 CSeq: 21 INVITE Server: Asterisk PBX 16.2.1~dfsg-1+deb10u2 Reason: Q.850;cause=34 l: 0 ``` The main thing that I can’t understand is that I see completely no output from ```auto % tcpdump -ni any host 91.121.209.194 or host 54.37.202.229 or host 212.79.111.155 ``` for `testB` call, while `testA` call produces it. If I ping or telnet [iptel.org](http://iptel.org) 5060, I see corresponding packages on tcpdump. It looks like asterisk doesn’t even try to connect to [iptel.org](http://iptel.org). Asterisk 16.2 debian stable ---
**Author:** ![david551](https://avatars.discourse-cdn.com/v4/letter/d/3bc359/32.png) [@david551](https://community.asterisk.org/u/david551)\ **Post date:** [September 29, 2020, 9:51am UTC](https://community.asterisk.org/t/resolving-issue/85861/2 "2020-09-29T09:51:01Z")
Normally, if you turn up the logging, there is a point at which you will see a message saying why the call failed. The Everyone is busy message is rather late in the process. ---
**Author:** ![sergio](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/sergio/32/4350_2.png) [@sergio](https://community.asterisk.org/u/sergio)\ **Post date:** [September 29, 2020, 12:19pm UTC](https://community.asterisk.org/t/resolving-issue/85861/3 "2020-09-29T12:19:01Z")
> the logging is `pjsip set logger on` enough? > The Everyone is busy message is rather late in the process The log before is the same in both cases with the exception of digits: ```auto <--- Received SIP request (1439 bytes) from TLS::42593 ---> INVITE sip:testB@domain.tld SIP/2.0 Via: SIP/2.0/TLS :42593;branch=;rport From: ;tag= To: sip:testB@domain.tld CSeq: 20 INVITE Call-ID: Max-Forwards: 70 Route: Supported: replaces, outbound, gruu Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE Content-Type: application/sdp Content-Length: 765 Contact: :42593;transport=tls>;expires=59;+sip.instance=">" User-Agent: LinphoneAndroid/4.3.1 (client) LinphoneSDK/4.4.2 (tags/4.4.2^0) v=0 o=client 396 2555 IN IP4 s=Talk c=IN IP4 t=0 0 a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics m=audio 7078 RTP/SAVP 96 0 8 9 18 101 97 a=rtpmap:96 opus/48000/2 a=fmtp:96 useinbandfec=1 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/48000 a=rtpmap:97 telephone-event/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:<> a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:<> a=crypto:3 AES_256_CM_HMAC_SHA1_80 inline:<> a=crypto:4 AES_256_CM_HMAC_SHA1_32 inline:<> a=rtcp-fb:* trr-int 1000 a=rtcp-fb:* ccm tmmbr <--- Transmitting SIP response (438 bytes) to TLS::42593 ---> SIP/2.0 401 Unauthorized v: SIP/2.0/TLS :42593;rport=42593;received=;branch= i: f: ;tag= t: ;tag= CSeq: 20 INVITE WWW-Authenticate: Digest realm="asterisk",nonce="",opaque="",algorithm=md5,qop="auth" Server: Asterisk PBX 16.2.1~dfsg-1+deb10u2 l: 0 <--- Received SIP request (456 bytes) from TLS::42593 ---> ACK sip:testB@domain.tld SIP/2.0 Via: SIP/2.0/TLS :42593;branch=;rport Call-ID: From: ;tag= To: ;tag= Contact: :42593;transport=tls>;expires=59;+sip.instance=">" Route: Max-Forwards: 70 CSeq: 20 ACK Content-Length: 0 <--- Received SIP request (1719 bytes) from TLS::42593 ---> INVITE sip:testB@domain.tld SIP/2.0 Via: SIP/2.0/TLS :42593;branch=;rport From: ;tag= To: sip:testB@domain.tld CSeq: 21 INVITE Call-ID: Max-Forwards: 70 Route: Supported: replaces, outbound, gruu Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE Content-Type: application/sdp Content-Length: 765 Contact: :42593;transport=tls>;expires=59;+sip.instance=">" User-Agent: LinphoneAndroid/4.3.1 (client) LinphoneSDK/4.4.2 (tags/4.4.2^0) Authorization: Digest realm="asterisk", nonce="", algorithm=md5, opaque="", username="client", uri="sip:testB@domain.tld", response="", cnonce="", nc=00000001, qop=auth v=0 o=client 396 2555 IN IP4 s=Talk c=IN IP4 t=0 0 a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics m=audio 7078 RTP/SAVP 96 0 8 9 18 101 97 a=rtpmap:96 opus/48000/2 a=fmtp:96 useinbandfec=1 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/48000 a=rtpmap:97 telephone-event/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:<> a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:<> a=crypto:3 AES_256_CM_HMAC_SHA1_80 inline:<> a=crypto:4 AES_256_CM_HMAC_SHA1_32 inline:<> a=rtcp-fb:* trr-int 1000 a=rtcp-fb:* ccm tmmbr == Setting global variable 'SIPDOMAIN' to 'domain.tld' <--- Transmitting SIP response (264 bytes) to TLS::42593 ---> SIP/2.0 100 Trying v: SIP/2.0/TLS :42593;rport=42593;received=;branch= i: f: ;tag= t: CSeq: 21 INVITE Server: Asterisk PBX 16.2.1~dfsg-1+deb10u2 l: 0 -- Executing [testB@context:1] Dial("PJSIP/client-0000000d", "PJSIP/outgoing/sip:thetestcall@iptel.org") in new stack ``` diff with test[AB] reduced to test: ```auto % diff /tmp/test* 18c18 < o=client 802 2454 IN IP4 --- > o=client 396 2555 IN IP4 79c79 < o=client 802 2454 IN IP4 --- > o=client 396 2555 IN IP4 109c109 < -- Executing [test@context:1] Dial("PJSIP/client-0000000f", "PJSIP/outgoing/sip:thetestcall@sip.linphone.org") in new stack --- > -- Executing [test@context:1] Dial("PJSIP/client-0000000d", "PJSIP/outgoing/sip:thetestcall@iptel.org") in new stack ``` ---
**Author:** ![david551](https://avatars.discourse-cdn.com/v4/letter/d/3bc359/32.png) [@david551](https://community.asterisk.org/u/david551)\ **Post date:** [September 29, 2020, 12:45pm UTC](https://community.asterisk.org/t/resolving-issue/85861/4 "2020-09-29T12:45:15Z")
No. You are looking for a warning, debugging or verbose message. The differences you are showing are, loosely, serial numbers. They are session numbers and SDP version numbers. ---
**Author:** ![sergio](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/sergio/32/4350_2.png) [@sergio](https://community.asterisk.org/u/sergio)\ **Post date:** [September 29, 2020, 1:07pm UTC](https://community.asterisk.org/t/resolving-issue/85861/5 "2020-09-29T13:07:35Z")
I added `debug` for `console`: ```auto console => notice,warning,error,debug ``` in `logger.conf` and looks like the issue is in domain resolution: linphone: ```auto res_pjsip/pjsip_resolver.c:477 sip_resolve: Performing SIP DNS resolution of target 'sip.linphone.org' res_pjsip/pjsip_resolver.c:504 sip_resolve: Transport type for target 'sip.linphone.org' is 'Unspecified' res_pjsip/pjsip_resolver.c:547 sip_resolve: [0x7f4e740593f8] Created resolution tracking for target 'sip.linphone.org' res_pjsip/pjsip_resolver.c:177 sip_resolve_add: [0x7f4e740593f8] Added target 'sip.linphone.org' with record type '35', transport 'Unspecified', and port '0' res_pjsip/pjsip_resolver.c:177 sip_resolve_add: [0x7f4e740593f8] Added target '_sips._tcp.sip.linphone.org' with record type '33', transport 'TLS', and port '5061' res_pjsip/pjsip_resolver.c:177 sip_resolve_add: [0x7f4e740593f8] Added target '_sip._tcp.sip.linphone.org' with record type '33', transport 'TCP', and port '5060' res_pjsip/pjsip_resolver.c:177 sip_resolve_add: [0x7f4e740593f8] Added target '_sip._udp.sip.linphone.org' with record type '33', transport 'UDP', and port '5060' res_pjsip/pjsip_resolver.c:616 sip_resolve: [0x7f4e740593f8] Starting initial resolution using parallel queries for target 'sip.linphone.org' res_pjsip_session.c:3538 session_inv_on_state_changed: Source of transaction state change is TX_MSG dns.c:557 ast_search_dns_ex: DNS search failed for sip.linphone.org dns_system_resolver.c:155 dns_system_resolver_process_query: DNS search failed for query: 'sip.linphone.org' res_pjsip/pjsip_resolver.c:277 sip_resolve_callback: [0x7f4e740593f8] All parallel queries completed res_pjsip/pjsip_resolver.c:349 sip_resolve_callback: [0x7f4e740593f8] SRV record received on target '_sips._tcp.sip.linphone.org' res_pjsip/pjsip_resolver.c:177 sip_resolve_add: [0x7f4e740593f8] Added target 'sip6.linphone.org' with record type '1', transport 'TLS', and port '5223' res_pjsip/pjsip_resolver.c:349 sip_resolve_callback: [0x7f4e740593f8] SRV record received on target '_sips._tcp.sip.linphone.org' res_pjsip/pjsip_resolver.c:177 sip_resolve_add: [0x7f4e740593f8] Added target 'sip1.linphone.org' with record type '1', transport 'TLS', and port '5223' res_pjsip/pjsip_resolver.c:349 sip_resolve_callback: [0x7f4e740593f8] SRV record received on target '_sips._tcp.sip.linphone.org' res_pjsip/pjsip_resolver.c:177 sip_resolve_add: [0x7f4e740593f8] Added target 'sip6.linphone.org' with record type '1', transport 'TLS', and port '443' res_pjsip/pjsip_resolver.c:349 sip_resolve_callback: [0x7f4e740593f8] SRV record received on target '_sip._tcp.sip.linphone.org' res_pjsip/pjsip_resolver.c:349 sip_resolve_callback: [0x7f4e740593f8] SRV record received on target '_sip._tcp.sip.linphone.org' res_pjsip/pjsip_resolver.c:349 sip_resolve_callback: [0x7f4e740593f8] SRV record received on target '_sip._udp.sip.linphone.org' res_pjsip/pjsip_resolver.c:349 sip_resolve_callback: [0x7f4e740593f8] SRV record received on target '_sip._udp.sip.linphone.org' res_pjsip/pjsip_resolver.c:413 sip_resolve_callback: [0x7f4e740593f8] New queries added, performing parallel resolution again res_pjsip/pjsip_resolver.c:277 sip_resolve_callback: [0x7f4e740593f8] All parallel queries completed res_pjsip/pjsip_resolver.c:326 sip_resolve_callback: [0x7f4e740593f8] A record received on target 'sip6.linphone.org' res_pjsip/pjsip_resolver.c:326 sip_resolve_callback: [0x7f4e740593f8] A record received on target 'sip1.linphone.org' res_pjsip/pjsip_resolver.c:326 sip_resolve_callback: [0x7f4e740593f8] A record received on target 'sip6.linphone.org' res_pjsip/pjsip_resolver.c:419 sip_resolve_callback: [0x7f4e740593f8] Resolution completed - 3 viable targets res_pjsip/pjsip_resolver.c:201 sip_resolve_invoke_user_callback: [0x7f4e740593f8] Address '0' is 54.37.202.229:5223 with transport 'TLS' res_pjsip/pjsip_resolver.c:201 sip_resolve_invoke_user_callback: [0x7f4e740593f8] Address '1' is 91.121.209.194:5223 with transport 'TLS' res_pjsip/pjsip_resolver.c:201 sip_resolve_invoke_user_callback: [0x7f4e740593f8] Address '2' is 54.37.202.229:443 with transport 'TLS' res_pjsip/pjsip_resolver.c:207 sip_resolve_invoke_user_callback: [0x7f4e740593f8] Invoking user callback with '3' addresses ``` iptel: ```auto res_pjsip/pjsip_resolver.c:477 sip_resolve: Performing SIP DNS resolution of target 'iptel.org' res_pjsip/pjsip_resolver.c:504 sip_resolve: Transport type for target 'iptel.org' is 'Unspecified' res_pjsip/pjsip_resolver.c:547 sip_resolve: [0x7f4e740564e8] Created resolution tracking for target 'iptel.org' res_pjsip/pjsip_resolver.c:177 sip_resolve_add: [0x7f4e740564e8] Added target 'iptel.org' with record type '35', transport 'Unspecified', and port '0' res_pjsip/pjsip_resolver.c:177 sip_resolve_add: [0x7f4e740564e8] Added target '_sips._tcp.iptel.org' with record type '33', transport 'TLS', and port '5061' res_pjsip/pjsip_resolver.c:177 sip_resolve_add: [0x7f4e740564e8] Added target '_sip._tcp.iptel.org' with record type '33', transport 'TCP', and port '5060' res_pjsip/pjsip_resolver.c:177 sip_resolve_add: [0x7f4e740564e8] Added target '_sip._udp.iptel.org' with record type '33', transport 'UDP', and port '5060' res_pjsip/pjsip_resolver.c:616 sip_resolve: [0x7f4e740564e8] Starting initial resolution using parallel queries for target 'iptel.org' res_pjsip_session.c:3538 session_inv_on_state_changed: Source of transaction state change is TX_MSG dns.c:557 ast_search_dns_ex: DNS search failed for iptel.org dns_system_resolver.c:155 dns_system_resolver_process_query: DNS search failed for query: 'iptel.org' dns.c:557 ast_search_dns_ex: DNS search failed for _sips._tcp.iptel.org dns_system_resolver.c:155 dns_system_resolver_process_query: DNS search failed for query: '_sips._tcp.iptel.org' res_pjsip/pjsip_resolver.c:277 sip_resolve_callback: [0x7f4e740564e8] All parallel queries completed res_pjsip/pjsip_resolver.c:349 sip_resolve_callback: [0x7f4e740564e8] SRV record received on target '_sip._tcp.iptel.org' res_pjsip/pjsip_resolver.c:349 sip_resolve_callback: [0x7f4e740564e8] SRV record received on target '_sip._udp.iptel.org' res_pjsip/pjsip_resolver.c:419 sip_resolve_callback: [0x7f4e740564e8] Resolution completed - 0 viable targets res_pjsip/pjsip_resolver.c:207 sip_resolve_invoke_user_callback: [0x7f4e740564e8] Invoking user callback with '0' addresses ``` but manually it works fine: ```auto % host iptel.org iptel.org has address 212.79.111.155 iptel.org mail is handled by 50 mx3.zoho.com. iptel.org mail is handled by 10 mx.zoho.com. iptel.org mail is handled by 20 mx2.zoho.com. % host -t SRV _sip._tcp.iptel.org _sip._tcp.iptel.org has SRV record 0 100 5060 sip.iptel.org. % host -t SRV _sip._udp.iptel.org _sip._udp.iptel.org has SRV record 0 25 5060 sip.iptel.org. % host sip.iptel.org. sip.iptel.org has address 212.79.111.155 ``` ---
**Author:** ![sergio](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/sergio/32/4350_2.png) [@sergio](https://community.asterisk.org/u/sergio)\ **Post date:** [September 29, 2020, 2:04pm UTC](https://community.asterisk.org/t/resolving-issue/85861/6 "2020-09-29T14:04:53Z")
I have another asterisk 16.3 run on OpenWrt 19.07.3 that uses the same local resolver and configured similarly, and it works fine with [iptel.org](http://iptel.org) ---
**Author:** ![sergio](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/sergio/32/4350_2.png) [@sergio](https://community.asterisk.org/u/sergio)\ **Post date:** [October 1, 2020, 1:02am UTC](https://community.asterisk.org/t/resolving-issue/85861/7 "2020-10-01T01:02:20Z")
It was due to a lack of tcp or udp sections with transport declaration in pjsip.conf. But it’s still unclear, 1. How should I find this? Is a log so poor and needs to be reported, or am I missing something? 2. Why I need to set bind? I use this transport only for outgoing connections. ---
**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\ **Post date:** [October 31, 2020, 1:02am UTC](https://community.asterisk.org/t/resolving-issue/85861/8 "2020-10-31T01:02:23Z")
Closed on October 31, 2020, 1:02am UTC This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.