SOLVED Grandstream GXP2xxx calls to VM or certain media get no audio

[wa4zlw]

my continued escapades with the latest incredible 13-12.2/SL 6.7

I installed the pbx today at the synagogue and started to have issues right off the bat. I finally got outbound calls working as well as extension-2-extension and I get two-way audio. There is a hardware watchguard firewall appliance in front of everything and we had the pbx here prior to the brief move to the cloud over the last 45 days or so. SO all my rules are still there that used to work.

inbound calls either go to the failover # (cell phone) or might come in but no audio is ever presented.

if I try 7777 from an extension same thing no audio but looking at the debugs it's actually working.

calls to VM does likewise.

also our old GXP2000s don't seem to do BLF or presence anymore yet those buttons dial the appropriate extension.

so I am stumped.

any suggestions welcome.

Thanks leon

root@pbx:~ $ status

Incredible PBX 13-12.2 for Scientific Linux

Asterisk: UP Apache: UP MySQL: UP
SendMail: UP IPtables: UP SSH: UP
LAN port: UP Fail2Ban: UP Webmin: UP
GV OAUTH: DN PortKnock: DN NR VPN: UP
FaxGetty: UP IAX Modem: UP HylaFax: UP

RAM:3317MB Scientific Linux 6.7 Disk:47GB

Asterisk 13.6.0 Incredible GUI 12.0.30

Private IP: 10.196.4.10 10.196.100.10

Public Info: 173.x.y.z

System Time: Sun Jan 17 14:59:39 EST 2016

< OK >

WARNING: Always run Incredible PBX behind a secure hardware-based firewall.
root@pbx:~ $

[EDITED] tried to post logs but greater than 10k is there an option to upload an attachment?

 

[wa4zlw]

ok the inbound calls were not be routed from vitelity or voip.ms now that they're getting to the pbx we're back to no audio coming back
like with 7777 or VM

 

[wa4zlw]

looks like my recorded announcements plus some of the system announcements are not there!

[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Executing [recordcheck@sub-record-check:1] NoOp("PJSIP/715-00000000", "Starting recording check against dontcare") in new stack
[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Executing [recordcheck@sub-record-check:2] Goto("PJSIP/715-00000000", "dontcare") in new stack
[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Goto (sub-record-check,recordcheck,3)
[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Executing [recordcheck@sub-record-check:3] Return("PJSIP/715-00000000", "") in new stack
[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Executing [in@sub-record-check:5] Return("PJSIP/715-00000000", "") in new stack
[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Executing [s@ext-did:6] Gosub("PJSIP/715-00000000", "app-blacklist-check,s,1()") in new stack
[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Executing [s@app-blacklist-check:1] GotoIf("PJSIP/715-00000000", "0?blacklisted") in new stack
[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Executing [s@app-blacklist-check:2] Set("PJSIP/715-00000000", "CALLED_BLACKLIST=1") in new stack
[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Executing [s@app-blacklist-check:3] Return("PJSIP/715-00000000", "") in new stack
[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Executing [s@ext-did:7] Set("PJSIP/715-00000000", "CDR(did)=7777") in new stack
[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Executing [s@ext-did:8] ExecIf("PJSIP/715-00000000", "0 ?Set(CALLERID(name)=715)") in new stack
[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Executing [s@ext-did:9] Set("PJSIP/715-00000000", "CHANNEL(musicclass)=default") in new stack
[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Executing [s@ext-did:10] Set("PJSIP/715-00000000", "__MOHCLASS=default") in new stack
[2016-01-17 15:49:40] WARNING[20626][C-00000000] func_channel.c: Unknown or unavailable item requested: 'reversecharge'
[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Executing [s@ext-did:11] GotoIf("PJSIP/715-00000000", "0?macro-hangupcall") in new stack
[2016-01-17 15:49:40] WARNING[20626][C-00000000] func_callerid.c: CALLERPRES is deprecated. Use CALLERID(name-pres) or CALLERID(num-pres) instead.
[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Executing [s@ext-did:12] Set("PJSIP/715-00000000", "__CALLINGPRES_SV=allowed_not_screened") in new stack
[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Executing [s@ext-did:13] Set("PJSIP/715-00000000", "CALLERPRES()=allowed_not_screened") in new stack
[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Executing [s@ext-did:14] Goto("PJSIP/715-00000000", "ivr-2,s,1") in new stack
[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Goto (ivr-2,s,1)
[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Executing [s@ivr-2:1] Set("PJSIP/715-00000000", "TIMEOUT_LOOPCOUNT=0") in new stack
[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Executing [s@ivr-2:2] Set("PJSIP/715-00000000", "INVALID_LOOPCOUNT=0") in new stack
[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Executing [s@ivr-2:3] Set("PJSIP/715-00000000", "_IVR_CONTEXT_ivr-2=") in new stack
[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Executing [s@ivr-2:4] Set("PJSIP/715-00000000", "_IVR_CONTEXT=ivr-2") in new stack
[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Executing [s@ivr-2:5] Set("PJSIP/715-00000000", "__IVR_RETVM=") in new stack
[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Executing [s@ivr-2:6] GotoIf("PJSIP/715-00000000", "0?skip") in new stack
[2016-01-17 15:49:40] VERBOSE[20626][C-00000000] pbx.c: Executing [s@ivr-2:7] Answer("PJSIP/715-00000000", "") in new stack
[2016-01-17 15:49:41] VERBOSE[20626][C-00000000] pbx.c: Executing [s@ivr-2:8] Wait("PJSIP/715-00000000", "1") in new stack
[2016-01-17 15:49:42] VERBOSE[20626][C-00000000] pbx.c: Executing [s@ivr-2:9] Set("PJSIP/715-00000000", "IVR_MSG=custom/shul-IVR-main") in new stack
[2016-01-17 15:49:42] VERBOSE[20626][C-00000000] pbx.c: Executing [s@ivr-2:10] Set("PJSIP/715-00000000", "TIMEOUT(digit)=3") in new stack
[2016-01-17 15:49:42] VERBOSE[20626][C-00000000] func_timeout.c: Digit timeout set to 3.000
[2016-01-17 15:49:42] VERBOSE[20626][C-00000000] pbx.c: Executing [s@ivr-2:11] ExecIf("PJSIP/715-00000000", "1?Background(custom/shul-IVR-main)") in new stack
[2016-01-17 15:49:42] WARNING[20626][C-00000000] file.c: File custom/shul-IVR-main does not exist in any format
[2016-01-17 15:49:42] WARNING[20626][C-00000000] file.c: Unable to open custom/shul-IVR-main (format (ulaw|g726)): No such file or directory
[2016-01-17 15:49:42] WARNING[20626][C-00000000] pbx.c: ast_streamfile failed on PJSIP/715-00000000 for custom/shul-IVR-main
[2016-01-17 15:49:42] VERBOSE[20626][C-00000000] pbx.c: Executing [s@ivr-2:12] WaitExten("PJSIP/715-00000000", "10,") in new stack

 

[wa4zlw]

I've rebuilt my announcements and it looks like other things are missing!

[2016-01-17 16:35:12] WARNING[24628] pbx.c: Context 'app-blacklist-last' tries to include nonexistent context 'app-blacklist-last-custom'
[2016-01-17 16:35:12] WARNING[24628] pbx.c: Context 'app-blacklist-check' tries to include nonexistent context 'app-blacklist-check-custom'
[2016-01-17 16:35:12] WARNING[24628] pbx.c: Context 'app-blacklist' tries to include nonexistent context 'app-blacklist-custom'
[2016-01-17 16:35:12] VERBOSE[24628] loader.c: Reloading module 'app_flite.so' (Flite TTS Interface)
[2016-01-17 16:35:12] WARNING[24628] app_flite.c: Flite: Unable to read config file flite.conf. Using default settings
[2016-01-17 16:35:12] VERBOSE[24628] loader.c: Reloading module 'codec_speex.so' (Speex Coder/Decoder)
[2016-01-17 16:35:12] VERBOSE[24628] loader.c: Reloading module 'app_amd.so' (Answering Machine Detection Application)

This is just a short snippet of what's displayed

 

[wa4zlw]

it looks like things are definitely missing in this install. While VMs are not working and other things is beyond me.

 

[wa4zlw]

some other weird results....my new GXP2170 I can get to VM and get audio. I can also dial the same extension and it rings. (All extensions are PJSIP here and have multiple deivces)

on the other devices they are GXP2000 and I can not get VM audio. Also, I found that on acct 1 ext 701 if I try and dial 701 from it I get a CALL FAILED REASON CODE: 603. on acct 2 which is x705 and acct 3 which is x710 i can dial same extensions and it rings. yet none of the accounts on the phone when I dial VM gets audio. THe display does indicate talking. There are three other (two configured) GXP2000 and they work the same.

This is totally weird. I do believe some things are missing due to what I see in the debug log but I am definitely seeing the right things happened especially when I dial VM. all these settings worked on Incredible 12-11 as well as VoipNow (Asterisk 11.7) including the soft keys and indicators.

I've spent the good part of today working on this. if any

 

[krzykat]

Why are you using PJSIP instead of SIP? I'd make things as simple as possible and then add to the equation afterwards.

 

[wa4zlw]

I have multiple extensions. I may create a new extension just to try it with only one device. I did try the same device on my digital ocean image and does the same thing. Looking at SNMP debug logs I am seeing when i dial *97 I'm getting unauthorized coming back. That makes no sense when I can dial on it. thoughts?

thanks leon

 

[wa4zlw]

I added SIP into the mix and found some interesting things:

1. I set sip to use 5068 and also the extension
2. extension shows up in chan sip peers
Chan_Sip Peers

Name/username Host Dyn Forcerport Comedia ACL Port Status Description
716 (Unspecified) D No No A 0 UNKNOWN
1 sip peers [Monitored: 0 online, 1 offline Unmonitored: 0 online, 0 offline]

BUT extension 99701 now shows up in pjsip:
Endpoint: 99701/99701 Unavailable 0 of inf
InAuth: 99701-auth/99701
Aor: 99701

restarted asterisk and device. this shows up for extension 716, note that PJSIP is reporting back NOT CHAN SIP. Also SYSLOG is showing Unauthorized on port 5068-->5068

[2016-01-18 09:18:48] NOTICE[4383] res_pjsip/pjsip_distributor.c: Request from '"Leon Test" <sip:[email protected]>' failed for '10.161.51.16:5068' (callid: [email protected]) - No matching endpoint found

If I change 716 to pjsip with only one extension everything works like the other extensions from the older Grandstream phones.

Leon

 

[wa4zlw]

I was able to pull data out of the SYSLOG:


9:57:44 10.161.51.16 <12>GS_LOG: [00:0B:82:1E:79:E0][720][9620000522A][0101062C] Packet Dropped During Provision: OPTIONS sip:[email protected]:5066 SIP/2.0 Via: SIP/2.0/UDP 10.196.4.10:5060;rport;branch=z9hG4bKPjbab86848-fb43-420f-9999-8d4b8f4fa875 From: <sip:[email protected]>;tag=134fd72e-d028-4372-8eeb-bf23c51863b0 To: <sip:[email protected]> Contact: <sip:[email protected]:5060> Call-ID: 1d8eaf78-a34b-4842-aded-cab3ed7d344d CSeq: 21547 OPTIONS Max-Forwards: 70 User-Agent: FPBX-12.0.70(13.6.0) Content-Length: 0

9:57:48 10.161.51.16 <12>GS_LOG: [00:0B:82:1E:79:E0][720][9620000522A][0101062C] Packet Dropped During Provision: OPTIONS sip:[email protected]:5066 SIP/2.0 Via: SIP/2.0/UDP 10.196.4.10:5060;rport;branch=z9hG4bKPjbab86848-fb43-420f-9999-8d4b8f4fa875 From: <sip:[email protected]>;tag=134fd72e-d028-4372-8eeb-bf23c51863b0 To: <sip:[email protected]> Contact: <sip:[email protected]:5060> Call-ID: 1d8eaf78-a34b-4842-aded-cab3ed7d344d CSeq: 21547 OPTIONS Max-Forwards: 70 User-Agent: FPBX-12.0.70(13.6.0) Content-Length: 0

10:00:48 10.161.51.16 <15>GS_LOG: [00:0B:82:1E:79:E0][000][9620000522A][0101062C] sip_len: 946, sip_handle: 0x0052F10A, INVITE sip:*[email protected] SIP/2.0 Via: SIP/2.0/UDP 10.161.51.16:5066;branch=z9hG4bK210cacce91de33cf From: "Leon Test" <sip:[email protected]>;tag=7a2ffd1fc73f5997 To: <sip:*[email protected]> Contact: <sip:[email protected]:5066;transport=udp> Supported: replaces, timer, 100rel, path Call-ID: [email protected] CSeq: 6 INVITE User-Agent: Grandstream GXP2000 1.1.6.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 403 v=0 o=716 8000 8000 IN IP4 10.161.51.16 s=SIP Call c=IN IP4 10.161.51.16 t=0 0 m=audio 5050 RTP/AVP 0 8 4 18 2 97 9 3 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11

10:00:48 10.161.51.16 <15>GS_LOG: [00:0B:82:1E:79:E0][000][9620000522A][0101062C] SIPReceive(498, Account4): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.161.51.16:5066;rport=5066;received=10.161.51.16;branch=z9hG4bK210cacce91de33cf Call-ID: [email protected] From: "Leon Test" <sip:[email protected]>;tag=7a2ffd1fc73f5997 To: <sip:*[email protected]>;tag=z9hG4bK210cacce91de33cf CSeq: 6 INVITE WWW-Authenticate: Digest realm="asterisk",nonce="1453129248/eb6f07369f9a6c70b51b5919272e602e",opaque="2b077baa58a1a4f2",algorithm=md5,qop="auth" Server: FPBX-12.0.70(13.6.0) Content-Length: 0

10:00:48 10.161.51.16 <15>GS_LOG: [00:0B:82:1E:79:E0][000][9620000522A][0101062C] sip_len: 507, sip_handle: 0x0052F10A, ACK sip:*[email protected] SIP/2.0 Via: SIP/2.0/UDP 10.161.51.16:5066;branch=z9hG4bK210cacce91de33cf From: "Leon Test" <sip:[email protected]>;tag=7a2ffd1fc73f5997 To: <sip:*[email protected]>;tag=z9hG4bK210cacce91de33cf Contact: <sip:[email protected]:5066;transport=udp> Supported: path Call-ID: [email protected] CSeq: 6 ACK User-Agent: Grandstream GXP2000 1.1.6.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0

10:00:48 10.161.51.16 <15>GS_LOG: [00:0B:82:1E:79:E0][000][9620000522A][0101062C] sip_len: 1219, sip_handle: 0x0052F10A, INVITE sip:*[email protected] SIP/2.0 Via: SIP/2.0/UDP 10.161.51.16:5066;branch=z9hG4bK15053d3efccd794f From: "Leon Test" <sip:[email protected]>;tag=7a2ffd1fc73f5997 To: <sip:*[email protected]> Contact: <sip:[email protected]:5066;transport=udp> Supported: replaces, timer, 100rel, path Authorization: Digest username="716", realm="asterisk", algorithm=MD5, uri="sip:*[email protected]", qop=auth, nc=00000001, cnonce="46003cced50f0387", opaque="2b077baa58a1a4f2", nonce="1453129248/eb6f07369f9a6c70b51b5919272e602e", response="c6f357fe305d1b9b60b755cc30b45918" Call-ID: [email protected] CSeq: 7 INVITE User-Agent: Grandstream GXP2000 1.1.6.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 403 v=0 o=716 8000 8001 IN IP4 10.161.51.16 s=SIP Call c=IN IP4 10.161.51.16 t=0 0 m=audio 5050 RTP/AVP 0 8 4 18 2 97 9 3 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11

10:00:48 10.161.51.16 <15>GS_LOG: [00:0B:82:1E:79:E0][000][9620000522A][0101062C] SIPReceive(317, Account4): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.161.51.16:5066;rport=5066;received=10.161.51.16;branch=z9hG4bK15053d3efccd794f Call-ID: [email protected] From: "Leon Test" <sip:[email protected]>;tag=7a2ffd1fc73f5997 To: <sip:*[email protected]> CSeq: 7 INVITE Server: FPBX-12.0.70(13.6.0) Content-Length: 0

10:00:48 10.161.51.16 <15>GS_LOG: [00:0B:82:1E:79:E0][000][9620000522A][0101062C] SIPReceive(853, Account4): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.161.51.16:5066;rport=5066;received=10.161.51.16;branch=z9hG4bK15053d3efccd794f Call-ID: [email protected] From: "Leon Test" <sip:[email protected]>;tag=7a2ffd1fc73f5997 To: <sip:*[email protected]>;tag=e89ffa72-ab1c-4c11-9274-c6da6b29d75d CSeq: 7 INVITE Server: FPBX-12.0.70(13.6.0) Contact: <sip:10.196.4.10:5060> Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER Supported: 100rel, timer, replaces, norefersub Content-Type: application/sdp Content-Length: 273 v=0 o=- 8000 8003 IN IP4 10.196.4.10 s=Asterisk c=IN IP4 10.196.4.10 t=0 0 m=audio 13036 RTP/AVP 0 3 2 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv

 

[wa4zlw]

From another extension on same device:

10:10:05 10.161.51.16 <15>GS_LOG: [00:0B:82:1E:79:E0][000][9620000522A][0101062C] SIPReceive(423, Account1): OPTIONS sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.196.4.10:5060;rport;branch=z9hG4bKPj77bb668d-9e04-41a4-87c9-defdf0c79196 From: <sip:[email protected]>;tag=d59321e4-b50e-4604-8f43-cddf635c3c47 To: <sip:[email protected]> Contact: <sip:[email protected]:5060> Call-ID: e3cd1788-1e49-4936-8875-2d645506c0b5 CSeq: 5617 OPTIONS Max-Forwards: 70 User-Agent: FPBX-12.0.70(13.6.0) Content-Length: 0
10:10:05 10.161.51.16 <15>GS_LOG: [00:0B:82:1E:79:E0][000][9620000522A][0101062C] sip_len: 539, sip_handle: 0x0052F0AA, SIP/2.0 200 OK Via: SIP/2.0/UDP 10.196.4.10:5060;rport;branch=z9hG4bKPj77bb668d-9e04-41a4-87c9-defdf0c79196 From: <sip:[email protected]>;tag=d59321e4-b50e-4604-8f43-cddf635c3c47 To: <sip:[email protected]>;tag=d42ad16a91809cb0 Call-ID: e3cd1788-1e49-4936-8875-2d645506c0b5 CSeq: 5617 OPTIONS User-Agent: Grandstream GXP2000 1.1.6.44 Contact: <sip:[email protected]:5060;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer, 100rel Content-Length: 0
10:10:28 10.161.51.16 <15>GS_LOG: [00:0B:82:1E:79:E0][000][9620000522A][0101062C] SIPReceive(423, Account4): OPTIONS sip:[email protected]:5066 SIP/2.0 Via: SIP/2.0/UDP 10.196.4.10:5060;rport;branch=z9hG4bKPj9160eac9-d785-446d-b649-e4f8c093af51 From: <sip:[email protected]>;tag=a3b2c975-48d0-434e-97cc-aa1bf7c2b88a To: <sip:[email protected]> Contact: <sip:[email protected]:5060> Call-ID: b3c34cfd-0c97-4f75-8830-b43f15910fbf CSeq: 9592 OPTIONS Max-Forwards: 70 User-Agent: FPBX-12.0.70(13.6.0) Content-Length: 0
10:10:28 10.161.51.16 <15>GS_LOG: [00:0B:82:1E:79:E0][000][9620000522A][0101062C] sip_len: 539, sip_handle: 0x0052F10A, SIP/2.0 200 OK Via: SIP/2.0/UDP 10.196.4.10:5060;rport;branch=z9hG4bKPj9160eac9-d785-446d-b649-e4f8c093af51 From: <sip:[email protected]>;tag=a3b2c975-48d0-434e-97cc-aa1bf7c2b88a To: <sip:[email protected]>;tag=0f8ba88c5bdb10ab Call-ID: b3c34cfd-0c97-4f75-8830-b43f15910fbf CSeq: 9592 OPTIONS User-Agent: Grandstream GXP2000 1.1.6.44 Contact: <sip:[email protected]:5066;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer, 100rel Content-Length: 0
10:10:46 10.161.51.16 <15>GS_LOG: [00:0B:82:1E:79:E0][000][9620000522A][0101062C] sip_len: 930, sip_handle: 0x0052F0AA, INVITE sip:*[email protected] SIP/2.0 Via: SIP/2.0/UDP 10.161.51.16:5060;branch=z9hG4bK58c9b0a20f92bad9 From: "Rabbi Lipsker" <sip:[email protected]>;tag=205bef35c7b24a42 To: <sip:*[email protected]> Contact: <sip:[email protected]:5060;transport=udp> Supported: replaces, timer, 100rel, path Call-ID: [email protected] CSeq: 11062 INVITE User-Agent: Grandstream GXP2000 1.1.6.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 379 v=0 o=701 8000 8000 IN IP4 10.161.51.16 s=SIP Call c=IN IP4 10.161.51.16 t=0 0 m=audio 5072 RTP/AVP 0 9 4 18 2 97 3 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11
10:10:46 10.161.51.16 <15>GS_LOG: [00:0B:82:1E:79:E0][000][9620000522A][0101062C] SIPReceive(506, Account1): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.161.51.16:5060;rport=5060;received=10.161.51.16;branch=z9hG4bK58c9b0a20f92bad9 Call-ID: [email protected] From: "Rabbi Lipsker" <sip:[email protected]>;tag=205bef35c7b24a42 To: <sip:*[email protected]>;tag=z9hG4bK58c9b0a20f92bad9 CSeq: 11062 INVITE WWW-Authenticate: Digest realm="asterisk",nonce="1453129846/a50e2ae7650e90ef716a6ae61b1083a0",opaque="3beba5533b9388b8",algorithm=md5,qop="auth" Server: FPBX-12.0.70(13.6.0) Content-Length: 0
10:10:46 10.161.51.16 <15>GS_LOG: [00:0B:82:1E:79:E0][000][9620000522A][0101062C] sip_len: 515, sip_handle: 0x0052F0AA, ACK sip:*[email protected] SIP/2.0 Via: SIP/2.0/UDP 10.161.51.16:5060;branch=z9hG4bK58c9b0a20f92bad9 From: "Rabbi Lipsker" <sip:[email protected]>;tag=205bef35c7b24a42 To: <sip:*[email protected]>;tag=z9hG4bK58c9b0a20f92bad9 Contact: <sip:[email protected]:5060;transport=udp> Supported: path Call-ID: [email protected] CSeq: 11062 ACK User-Agent: Grandstream GXP2000 1.1.6.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0
10:10:46 10.161.51.16 <15>GS_LOG: [00:0B:82:1E:79:E0][000][9620000522A][0101062C] sip_len: 1203, sip_handle: 0x0052F0AA, INVITE sip:*[email protected] SIP/2.0 Via: SIP/2.0/UDP 10.161.51.16:5060;branch=z9hG4bK0c67d3ae35304230 From: "Rabbi Lipsker" <sip:[email protected]>;tag=205bef35c7b24a42 To: <sip:*[email protected]> Contact: <sip:[email protected]:5060;transport=udp> Supported: replaces, timer, 100rel, path Authorization: Digest username="701", realm="asterisk", algorithm=MD5, uri="sip:*[email protected]", qop=auth, nc=00000001, cnonce="19199a191c0cd35e", opaque="3beba5533b9388b8", nonce="1453129846/a50e2ae7650e90ef716a6ae61b1083a0", response="96aedfea093b66990a9ca87e95756b11" Call-ID: [email protected] CSeq: 11063 INVITE User-Agent: Grandstream GXP2000 1.1.6.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 379 v=0 o=701 8000 8001 IN IP4 10.161.51.16 s=SIP Call c=IN IP4 10.161.51.16 t=0 0 m=audio 5072 RTP/AVP 0 9 4 18 2 97 3 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11
10:10:46 10.161.51.16 <15>GS_LOG: [00:0B:82:1E:79:E0][000][9620000522A][0101062C] SIPReceive(325, Account1): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.161.51.16:5060;rport=5060;received=10.161.51.16;branch=z9hG4bK0c67d3ae35304230 Call-ID: [email protected] From: "Rabbi Lipsker" <sip:[email protected]>;tag=205bef35c7b24a42 To: <sip:*[email protected]> CSeq: 11063 INVITE Server: FPBX-12.0.70(13.6.0) Content-Length: 0
10:10:46 10.161.51.16 <15>GS_LOG: [00:0B:82:1E:79:E0][000][9620000522A][0101062C] SIPReceive(861, Account1): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.161.51.16:5060;rport=5060;received=10.161.51.16;branch=z9hG4bK0c67d3ae35304230 Call-ID: [email protected] From: "Rabbi Lipsker" <sip:[email protected]>;tag=205bef35c7b24a42 To: <sip:*[email protected]>;tag=d6888542-740b-4fdf-bae1-0f2e6f27474f CSeq: 11063 INVITE Server: FPBX-12.0.70(13.6.0) Contact: <sip:10.196.4.10:5060> Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER Supported: 100rel, timer, replaces, norefersub Content-Type: application/sdp Content-Length: 273 v=0 o=- 8000 8003 IN IP4 10.196.4.10 s=Asterisk c=IN IP4 10.196.4.10 t=0 0 m=audio 12442 RTP/AVP 0 3 2 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv
10:10:46 10.161.51.16 <15>GS_LOG: [00:0B:82:1E:79:E0][000][9620000522A][0101062C] sip_len: 802, sip_handle: 0x0052F0AA, ACK sip:10.196.4.10:5060 SIP/2.0 Via: SIP/2.0/UDP 10.161.51.16:5060;branch=z9hG4bKbfac10ad412d9afc From: "Rabbi Lipsker" <sip:[email protected]>;tag=205bef35c7b24a42 To: <sip:*[email protected]>;tag=d6888542-740b-4fdf-bae1-0f2e6f27474f Contact: <sip:[email protected]:5060;transport=udp> Supported: path Authorization: Digest username="701", realm="asterisk", algorithm=MD5, uri="sip:*[email protected]", qop=auth, nc=00000001, cnonce="19199a191c0cd35e", opaque="3beba5533b9388b8", nonce="1453129846/a50e2ae7650e90ef716a6ae61b1083a0", response="96aedfea093b66990a9ca87e95756b11" Call-ID: [email protected] CSeq: 11063 ACK User-Agent: Grandstream GXP2000 1.1.6.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0

 

[wa4zlw]

more details...it seems it is not related to just VM. some announcements looks like related to certain media streams.

 

[wa4zlw]

I fixed the problem with the media streams and the older phones. Thanks to Wireshark I found that the pbx was sending me GSM codec with no data rate. Phone was sending g711 and I was able to hear the stream from my phone. What I did was forced g711u on the pbx for all the extensions and now everything works.

I predict this is definitely related to Asterisk 13 as we never had this on Asterisk 11.

Being a Grandstream Reseller I've opened a support ticket on the BLF/indicator