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Thread: Caller ID when redirecting incoming call
hol

Replies: 0
Views: 542

Caller ID when redirecting incoming call 23.01.2019 15:37 Forum: Miscellaneous

Hi

My SIP trunk supplier now takes the caller ID from PBXes, allowing it to be configured rather than forcing a fixed ID.

I can't figure out how to get PBXes to put the line's caller ID when calling out from the office, but to pass on the incoming caller ID when redirecting in incoming call back out to mobiles.

If I put a caller ID on at the trunk level it always sends that. Is it possible to add it at extension level?


####################################
UPDATE
####################################

I should have looked more carefully at the extension settings before reaching out!

By putting the caller ID on each of the extensions and not putting it on the trunk the system works as I wanted it to - Our line caller id when calling out and the incoming id when redirecting back out to mobiles. Sorted.

Thread: Problem coming our of 'on-hold'
hol

Replies: 2
Views: 1407

Daumen hoch! RE: Problem coming our of 'on-hold' 23.01.2019 14:14 Forum: Terminal Equipment

I have disabled all codecs except G.722 and that has cured the problem. Thanks for your pointer, it was very helpful.

Thread: Trunk configuration changing
hol

Replies: 1
Views: 631

Trunk configuration changing 14.12.2018 16:04 Forum: Bugs

Getting very weird things happen when I update trunk configuration.

1) Select trunk to edit from list on left, config looks ok
2) Make changes, click save - SIP server URL changes to nothing I entered and some dialing rules appear
3) If I apply the changes the fields are normal once screen refreshes

Almost looks like a db corruption.

Tried logging out and in, but issue persists.

Any ideas?

Thread: Problem coming our of 'on-hold'
hol

Replies: 2
Views: 1407

Problem coming off of 'on-hold' 14.12.2018 13:11 Forum: Terminal Equipment

If I place a call on hold the caller hears music, but when coming off hold it sounds like white noise.

Link to recording of problem, putting call on hold: [URL]https://drive.google.com/open?id=1COiiHxdgxS87wBb5YLO9geschocktVnd_FL3WPj[/URL]

Can you help?

Thread: Doorbird - Audio not routing
hol

Replies: 15
Views: 3446

RE: Doorbird - Audio not routing 15.10.2018 17:55 Forum: Terminal Equipment

Thank you - that works. Your timing was brilliant, as I'm set to install the unit tomorrow and thought I was going to have to explain it only calling 1 extension directly.

Thread: Doorbird - Audio not routing
hol

Replies: 15
Views: 3446

RE: Doorbird - Audio not routing 15.10.2018 12:39 Forum: Terminal Equipment

Thanks for this. I will get back to the manufacturer.

Is there anything we can do in the meantime to ensure this error does not trip up PBXes? When I connect directly to phones they all seem to cope with it.

Thread: Doorbird - Audio not routing
hol

Replies: 15
Views: 3446

RE: Doorbird - Audio not routing 12.10.2018 19:40 Forum: Terminal Equipment

OK, just done a test call ~ 2018-10-12 18.39UTC

Thread: Doorbird - Audio not routing
hol

Replies: 15
Views: 3446

RE: Doorbird - Audio not routing 08.10.2018 10:06 Forum: Terminal Equipment

Thanks.

Just done another test call. 2018-10-08 at 09.04 UTC

Thread: Problem with classic extensions in ring groups
hol

Replies: 4
Views: 1379

RE: Problem with classic extensions in ring groups 10.09.2018 10:57 Forum: Miscellaneous

OK. Before I request the trunk provider to turn off the forced caller ID do I need to do anything in PBXes to make sure the normal outgoing calls have the right caller IDs on them?

Thread: Doorbird - Audio not routing
hol

Replies: 15
Views: 3446

RE: Audio not routing 10.09.2018 10:54 Forum: Terminal Equipment

Hi Pascal. Thanks for continuing investigations.

I added a MoH track and tested again (~09:52 UTC) and no apparent change - I certainly don't hear music at either end

13/9 Any update on this please?

Thread: Doorbird - Audio not routing
hol

Replies: 15
Views: 3446

RE: Audio not routing 07.09.2018 10:58 Forum: Terminal Equipment

Just did a test call (~ 09:58 UTC). Same issue - audio drops out after a second, video fine

Thread: Doorbird - Audio not routing
hol

Replies: 15
Views: 3446

RE: Audio not routing 03.09.2018 11:36 Forum: Terminal Equipment

Hi. Thanks for this offer.

Would either 1pm UK time (2pm DE) today 3rd August or 10am UK (11am DE) tomorrow 4th be convenient?

Thread: Doorbird - Audio not routing
hol

Replies: 15
Views: 3446

Audio not routing 20.08.2018 12:17 Forum: Terminal Equipment

Still trying to get Doorbird door entry phone to work.

If I configure it to call a local extension directly it works with no problem, video and audio route. If I go via PBXes the video routes but not the audio.

In both cases the phone reports a PMCU audio connection, but through PBXes the audio from Doorbird to phone extension reports 0bps dataa rate.

Any ideas?

Extract from SIP Trace below . Doorbird is holbrook-601 (IP 192.168.8.4) calling holbrook-105 (IP 192.168.8.10cool

Aug 20 11:53:14 VERBOSE[22242] logger.c:
<-- SIP read
INVITE sip:105@pbxes.com SIP/2.0
Via: SIP/2.0/UDP 192.168.8.4:5060;rport;branch=z9hG4bKPj2c4a573c-318d-4e16-9315-0d78e4537e3d
Max-Forwards: 70
From: sip:holbrook-601@pbxes.org;tag=1b3da137-865c-4eae-89ad-f1f723856236
P-src-ip: 86.152.199.14
To: sip:105@pbxes.com
Contact: <sip:holbrook-601@192.168.8.4:5060;ob>
Call-ID: fbb14a75-e672-4af1-8965-fab2ad098539
CSeq: 12536 INVITE
Call-Info: <http://192.168.8.4/bha-api/image.cgi> ;purpose=icon, <http://192.168.8.4/bha-api/> ;purpose=info
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Proxy-Authorization: Digest username="holbrook-601", realm="pbxes.org", nonce="1aa70fb81f30e1486a15168c4cbb5470753ffd57", uri="sip:105@pbxes.com", response="6e85558706480f20ddbf6ddae48b2617"
Content-Type: application/sdp
Content-Length: 541
v=0
o=- 3743751201 3743751201 IN IP4 192.168.8.4
s=doorbird
b=AS:84
t=0 0
a=X-nat:0
m=audio 4012 RTP/AVP 0 8 96
c=IN IP4 192.168.8.4
b=TIAS:64000
a=rtcp:4013 IN IP4 192.168.8.4
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
m=video 4014 RTP/AVP 97
c=IN IP4 192.168.8.4
a=rtcp:4015 IN IP4 192.168.8.4
a=sendonly
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42e01e; packetization-mode=1; sprop-parameter-sets=Z00AKNoBQBbpUgAABPAAAGLAwIAB6EgACJVF73wvCIRq,aO48gA==
Aug 20 11:53:14 VERBOSE[22242] logger.c: --- (17 headers 21 lines)Aug 20 11:53:14 VERBOSE[22242] logger.c: --- (17 headers 21 lines)---
Aug 20 11:53:14 VERBOSE[22242] logger.c: Using INVITE request as basis request - fbb14a75-e672-4af1-8965-fab2ad098539
Aug 20 11:53:14 VERBOSE[22242] logger.c: Found user 'holbrook-601'
Aug 20 11:53:14 VERBOSE[22242] logger.c: Found RTP audio format 0
Aug 20 11:53:14 VERBOSE[22242] logger.c: Found RTP audio format 8
Aug 20 11:53:14 VERBOSE[22242] logger.c: Found RTP audio format 96
Aug 20 11:53:14 VERBOSE[22242] logger.c: Found RTP video format 97
Aug 20 11:53:14 VERBOSE[22242] logger.c: Peer audio RTP is at port 192.168.8.4:4012
Aug 20 11:53:14 VERBOSE[22242] logger.c: Peer video RTP is at port 192.168.8.4:4014
Aug 20 11:53:14 VERBOSE[22242] logger.c: Found description format PCMU
Aug 20 11:53:14 VERBOSE[22242] logger.c: Found description format PCMA
Aug 20 11:53:14 VERBOSE[22242] logger.c: Found description format telephone-event
Aug 20 11:53:14 VERBOSE[22242] logger.c: Found description format H264
Aug 20 11:53:14 VERBOSE[22242] logger.c: Capabilities: us - 0x38161e (gsm|ulaw|alaw|g726|speex|ilbc|g722|h263|h263p|h264), peer - audio=0xc (ulaw|alaw)/video=0x200000 (h264), combined - 0x20000c (ulaw|alaw|h264)
Aug 20 11:53:14 VERBOSE[22242] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Aug 20 11:53:14 VERBOSE[22242] logger.c: Looking for 105 in from-internal (domain pbxes.com)
Aug 20 11:53:14 VERBOSE[22242] logger.c: list_route: hop: <sip:holbrook-601@192.168.8.4:5060;ob>
Aug 20 11:53:14 VERBOSE[22242] logger.c: Transmitting (NAT)
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.8.4:5060;branch=z9hG4bKPj2c4a573c-318d-4e16-9315-0d78e4537e3d;received=144.76.38.78;rport=49530
From: sip:holbrook-601@pbxes.org;tag=1b3da137-865c-4eae-89ad-f1f723856236
To: sip:105@pbxes.com
Call-ID: fbb14a75-e672-4af1-8965-fab2ad098539
CSeq: 12536 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:105@144.76.38.78:27756>
Content-Length: 0
---
Aug 20 11:53:15 VERBOSE[8529] logger.c: We're at 144.76.38.78 port 41968
Aug 20 11:53:15 VERBOSE[8529] logger.c: Video is at 144.76.38.78 port 37822
Aug 20 11:53:15 VERBOSE[22242] logger.c: 12 headers, 3 lines
Aug 20 11:53:15 VERBOSE[22242] logger.c: Reliably Transmitting (NAT)
NOTIFY sip:holbrook-601@192.168.8.4:5060;ob SIP/2.0
Via: SIP/2.0/UDP 144.76.38.78:27756;branch=z9hG4bK78655d48;rport
From: "Unknown" <sip:Unknown@144.76.38.78:27756>;tag=as29e835e6
To: <sip:holbrook-601@192.168.8.4:5060;ob>
Contact: <sip:Unknown@144.76.38.78:27756>
Call-ID: 63bffc382f3c520d5624540b6ffa088b@144.76.38.78
CSeq: 102 NOTIFY
User-Agent: PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 87
Messages-Waiting: no
Message-Account: sip:*97@144.76.38.78
Voice-Message: 0/0 (0/0)
---
Aug 20 11:53:15 VERBOSE[8529] logger.c: Adding codec 0x4 (ulaw) to SDP
Aug 20 11:53:15 VERBOSE[8529] logger.c: Adding codec 0x1000 (g722) to SDP
Aug 20 11:53:15 VERBOSE[8529] logger.c: Adding codec 0x8 (alaw) to SDP
Aug 20 11:53:15 VERBOSE[8529] logger.c: Adding codec 0x10 (g726) to SDP
Aug 20 11:53:15 VERBOSE[22242] logger.c: Scheduling destruction of call '63bffc382f3c520d5624540b6ffa088b@144.76.38.78' in 15000 ms
Aug 20 11:53:15 VERBOSE[8529] logger.c: Adding codec 0x400 (ilbc) to SDP
Aug 20 11:53:15 VERBOSE[8529] logger.c: Adding codec 0x200 (speex) to SDP
Aug 20 11:53:15 VERBOSE[8529] logger.c: Adding codec 0x2 (gsm) to SDP
Aug 20 11:53:15 VERBOSE[8529] logger.c: Adding codec 0x100000 (h263p) to SDP
Aug 20 11:53:15 VERBOSE[8529] logger.c: Adding codec 0x80000 (h263) to SDP
Aug 20 11:53:15 VERBOSE[8529] logger.c: Adding codec 0x200000 (h264) to SDP
Aug 20 11:53:15 VERBOSE[8529] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Aug 20 11:53:15 VERBOSE[8529] logger.c: 14 headers, 19 lines
Aug 20 11:53:15 VERBOSE[8529] logger.c: Reliably Transmitting (NAT)
INVITE sip:holbrook-105@192.168.8.108:5060 SIP/2.0
Via: SIP/2.0/UDP 144.76.38.78:27756;branch=z9hG4bK09ab8320;rport
From: "Front Door" <sip:601@144.76.38.78:27756>;tag=as09401853
To: <sip:holbrook-105@192.168.8.108:5060>
Contact: <sip:601@144.76.38.78:27756>
Call-ID: 02a6979c2ed9277e74c534130ff64b2d@144.76.38.78
CSeq: 102 INVITE
User-Agent: PBX
Max-Forwards: 70
Date: Mon, 20 Aug 2018 10:53:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
P-src-ip: 86.152.199.14
Content-Type: application/sdp
Content-Length: 457
v=0
o=root 12625 12625 IN IP4 144.76.38.78
s=session
c=IN IP4 144.76.38.78
t=0 0
m=audio 41968 RTP/AVP 0 9 8 111 97 110 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 37822 RTP/AVP 103 34 99
a=rtpmap:103 h263-1998/90000
a=rtpmap:34 H263/90000
a=rtpmap:99 H264/90000
---
Aug 20 11:53:15 VERBOSE[8529] logger.c: -- Called holbrook-105
Aug 20 11:53:15 VERBOSE[22242] logger.c:
<-- SIP read
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 144.76.38.78:5060;branch=z9hG4bK09ab8320;rport=5060
From: "Front Door" <sip:601@144.76.38.78:27756>;tag=as09401853
P-src-ip: 86.152.199.14
To: <sip:holbrook-105@192.168.8.108:5060>
Call-ID: 02a6979c2ed9277e74c534130ff64b2d@144.76.38.78
CSeq: 102 INVITE
Supported: replaces, path, eventlist
User-Agent: Grandstream GXV3275 1.0.3.198
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
Aug 20 11:53:15 VERBOSE[22242] logger.c: --- (11 headers 0 lines)Aug 20 11:53:15 VERBOSE[22242] logger.c: --- (11 headers 0 lines)---
Aug 20 11:53:15 VERBOSE[22242] chan_sip.c: SIP response 100 to standard invite
Aug 20 11:53:15 VERBOSE[22242] logger.c:
<-- SIP read
SIP/2.0 200 OK
Via: SIP/2.0/UDP 144.76.38.78:5060;rport=5060;received=144.76.38.78;branch=z9hG4bK78655d48
Call-ID: 63bffc382f3c520d5624540b6ffa088b@144.76.38.78
From: "Unknown" <sip:Unknown@144.76.38.78>;tag=as29e835e6
P-src-ip: 86.152.199.14
To: <sip:holbrook-601@192.168.8.4;ob>;tag=z9hG4bK78655d48
CSeq: 102 NOTIFY
Content-Length: 0
Aug 20 11:53:15 VERBOSE[22242] logger.c: --- (8 headers 0 lines)Aug 20 11:53:15 VERBOSE[22242] logger.c: --- (8 headers 0 lines)---
Aug 20 11:53:15 VERBOSE[22242] logger.c: Destroying call '63bffc382f3c520d5624540b6ffa088b@144.76.38.78'
Aug 20 11:53:15 VERBOSE[22242] logger.c:
<-- SIP read
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 144.76.38.78:5060;branch=z9hG4bK09ab8320;rport=5060
From: "Front Door" <sip:601@144.76.38.78:27756>;tag=as09401853
P-src-ip: 86.152.199.14
To: <sip:holbrook-105@192.168.8.108:5060>;tag=2104099985
Call-ID: 02a6979c2ed9277e74c534130ff64b2d@144.76.38.78
CSeq: 102 INVITE
Contact: <sip:holbrook-105@192.168.8.108:5060>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3275 1.0.3.198
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
Aug 20 11:53:15 VERBOSE[22242] logger.c: --- (13 headers 0 lines)Aug 20 11:53:15 VERBOSE[22242] logger.c: --- (13 headers 0 lines)---
Aug 20 11:53:15 VERBOSE[22242] chan_sip.c: SIP response 180 to standard invite
Aug 20 11:53:15 VERBOSE[8529] logger.c: -- SIP/holbrook-105-1ff6 is ringing
Aug 20 11:53:15 VERBOSE[8529] logger.c: Transmitting (NAT)
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.8.4:5060;branch=z9hG4bKPj2c4a573c-318d-4e16-9315-0d78e4537e3d;received=144.76.38.78;rport=49530
From: sip:holbrook-601@pbxes.org;tag=1b3da137-865c-4eae-89ad-f1f723856236
To: sip:105@pbxes.com;tag=as0a404948
Call-ID: fbb14a75-e672-4af1-8965-fab2ad098539
CSeq: 12536 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:105@144.76.38.78:27756>
Content-Length: 0
---
Aug 20 11:53:18 VERBOSE[22242] logger.c:
<-- SIP read
SIP/2.0 200 OK
Via: SIP/2.0/UDP 144.76.38.78:5060;branch=z9hG4bK09ab8320;rport=5060
From: "Front Door" <sip:601@144.76.38.78:27756>;tag=as09401853
P-src-ip: 86.152.199.14
To: <sip:holbrook-105@192.168.8.108:5060>;tag=2104099985
Call-ID: 02a6979c2ed9277e74c534130ff64b2d@144.76.38.78
CSeq: 102 INVITE
Contact: <sip:holbrook-105@192.168.8.108:5060>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3275 1.0.3.198
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 410
v=0
o=holbrook-105 8000 8000 IN IP4 192.168.8.108
s=SIP Call
c=IN IP4 192.168.8.108
t=0 0
m=audio 5004 RTP/AVP 0 8 9 101
a=sendrecv
a=rtcp:5005 IN IP4 192.168.8.108
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 0 RTP/AVP 103 34 99
a=rtpmap:103 h263-1998/90000
a=rtpmap:34 H263/90000
a=rtpmap:99 H264/90000
Aug 20 11:53:18 VERBOSE[22242] logger.c: --- (13 headers 18 lines)Aug 20 11:53:18 VERBOSE[22242] logger.c: --- (13 headers 18 lines)---
Aug 20 11:53:18 VERBOSE[22242] chan_sip.c: SIP response 200 to standard invite
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found RTP audio format 0
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found RTP audio format 8
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found RTP audio format 9
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found RTP audio format 101
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found RTP video format 103
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found RTP video format 34
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found RTP video format 99
Aug 20 11:53:18 VERBOSE[22242] logger.c: Peer audio RTP is at port 192.168.8.108:5004
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found description format PCMU
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found description format PCMA
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found description format G722
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found description format telephone-event
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found description format h263-1998
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found description format H263
Aug 20 11:53:18 VERBOSE[22242] logger.c: Found description format H264
Aug 20 11:53:18 VERBOSE[22242] logger.c: Capabilities: us - 0x38161e (gsm|ulaw|alaw|g726|speex|ilbc|g722|h263|h263p|h264), peer - audio=0x100c (ulaw|alaw|g722)/video=0x380000 (h263|h263p|h264), combined - 0x38100c (ulaw|alaw|g722|h263|h263p|h264)
Aug 20 11:53:18 VERBOSE[22242] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Aug 20 11:53:18 VERBOSE[22242] logger.c: list_route: hop: <sip:holbrook-105@192.168.8.108:5060>
Aug 20 11:53:18 VERBOSE[22242] logger.c: set_destination: Parsing <sip:holbrook-105@192.168.8.108:5060> for address/port to send to
Aug 20 11:53:18 VERBOSE[22242] logger.c: set_destination: set destination to 192.168.8.108, port 5060
Aug 20 11:53:18 VERBOSE[22242] logger.c: Transmitting (NAT)
ACK sip:holbrook-105@192.168.8.108:5060 SIP/2.0
Via: SIP/2.0/UDP 144.76.38.78:27756;branch=z9hG4bK112ecf2d;rport
From: "Front Door" <sip:601@144.76.38.78:27756>;tag=as09401853
To: <sip:holbrook-105@192.168.8.108:5060>;tag=2104099985
Contact: <sip:601@144.76.38.78:27756>
Call-ID: 02a6979c2ed9277e74c534130ff64b2d@144.76.38.78
CSeq: 102 ACK
User-Agent: PBX
Max-Forwards: 70
Content-Length: 0
---

Thread: Problem with classic extensions in ring groups
hol

Replies: 4
Views: 1379

RE: Problem with classic extensions in ring groups 16.08.2018 11:57 Forum: Miscellaneous

I have tried that, but it is still skipping this stage.

I am using the local phone umber format (e.g. 07811111111# ) but it is not working

2018-09-07 Update:
I found I could get the routing to work if I defined a Default outbound route. It looks like the routing logic is not respecting the Ext rules set up (I have mine in blocks of 100 for different companies/departments in a building). Even though the secondary ring group is in the range for the pattern match, it is not being matched.

So, a default route picks it up and sends the calls on - either to Classic extensions or numbers listed. This is not ideal, as only 1 outgoing line is used for different companies in the building.

Also, on routing to an outside number, the original caller ID is lost and replaced with the outgoing line's caller id. Is this something PBXes is doing or my SIP trunk provider?

Thread: Video compatibility problem?
hol

Replies: 2
Views: 1168

RE: Video compatibility problem? 15.08.2018 18:21 Forum: Miscellaneous

Thanks, that was very helpful - and worked, at least partially. I get the video through but no audio at the moment.

Thread: Problem with classic extensions in ring groups
hol

Replies: 4
Views: 1379

Problem with classic extensions in ring groups 15.08.2018 17:53 Forum: Miscellaneous

Our incoming calls go to a ring group 100 - calling all of 4 office extensions.

If not answered it goes to another group (150), but if this group includes classic extensions (ring all or hunt) the group is skipped and it goes straight to the next stage.

I can call ring group 150 from an extension and it works fine. I have checked and double checked there are no rogue extensions in any ring groups, so am puzzled as to why this is failing.

I have also found that if I select a classic extension if unanswered that also fails.

Scanning through the forum I can see some similar issues back in 2013, but I assume this is not a common problem.

Any ideas?

Thread: Video compatibility problem?
hol

Replies: 2
Views: 1168

Video compatibility problem? 14.08.2018 13:33 Forum: Miscellaneous

Trying to link a Doorbird door entry system into pbxes with partial success. I can (video) call the doorbird from a phone but not the other way around (which is how it should usually work!)

I am trying to call another extension using holbrook-xxx@pbxes.com as the sip address. I can see the status change to red briefly on the doorbird extension, but it cannot connect to the called extension.

extract from log:
Aug 14 13:44:06 VERBOSE[22242] logger.c: Found RTP audio format 0
Aug 14 13:44:06 VERBOSE[22242] logger.c: Found RTP audio format 8
Aug 14 13:44:06 VERBOSE[22242] logger.c: Found RTP audio format 96
Aug 14 13:44:06 VERBOSE[22242] logger.c: Found RTP video format 97
Aug 14 13:44:06 VERBOSE[22242] logger.c: Peer audio RTP is at port 192.168.8.4:4020
Aug 14 13:44:06 VERBOSE[22242] logger.c: Peer video RTP is at port 192.168.8.4:4022
Aug 14 13:44:06 VERBOSE[22242] logger.c: Found description format PCMU
Aug 14 13:44:06 VERBOSE[22242] logger.c: Found description format PCMA
Aug 14 13:44:06 VERBOSE[22242] logger.c: Found description format telephone-event
Aug 14 13:44:06 VERBOSE[22242] logger.c: Found description format H264
Aug 14 13:44:06 VERBOSE[22242] logger.c: Capabilities: us - 0x38161e (gsm|ulaw|alaw|g726|speex|ilbc|g722|h263|h263p|h264), peer - audio=0xc (ulaw|alaw)/video=0x200000 (h264), combined - 0x20000c (ulaw|alaw|h264)
Aug 14 13:44:06 VERBOSE[22242] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Aug 14 13:44:06 VERBOSE[33085] logger.c: We're at 144.76.38.78 port 43580
Aug 14 13:44:06 VERBOSE[33085] logger.c: Video is at 144.76.38.78 port 38142
Aug 14 13:44:06 VERBOSE[33085] logger.c: Adding codec 0x4 (ulaw) to SDP
Aug 14 13:44:06 VERBOSE[33085] logger.c: Adding codec 0x1000 (g722) to SDP
Aug 14 13:44:06 VERBOSE[33085] logger.c: Adding codec 0x8 (alaw) to SDP
Aug 14 13:44:06 VERBOSE[33085] logger.c: Adding codec 0x10 (g726) to SDP
Aug 14 13:44:06 VERBOSE[33085] logger.c: Adding codec 0x400 (ilbc) to SDP
Aug 14 13:44:06 VERBOSE[33085] logger.c: Adding codec 0x200 (speex) to SDP
Aug 14 13:44:06 VERBOSE[33085] logger.c: Adding codec 0x2 (gsm) to SDP
Aug 14 13:44:06 VERBOSE[33085] logger.c: Adding codec 0x100000 (h263p) to SDP
Aug 14 13:44:06 VERBOSE[33085] logger.c: Adding codec 0x80000 (h263) to SDP
Aug 14 13:44:06 VERBOSE[33085] logger.c: Adding codec 0x200000 (h264) to SDP
Aug 14 13:44:06 VERBOSE[33085] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Aug 14 13:44:06 VERBOSE[33085] logger.c: -- Called holbrook-105@pbxes.com
Aug 14 13:44:07 VERBOSE[22242] logger.c: -- Got SIP response 482 "Loop Detected"
Aug 14 13:44:07 VERBOSE[33085] logger.c: -- Now forwarding SIP/holbrook-601-31d4 to 'Local/holbrook-105@from-pstn/n' (thanks to SIP/pbxes.com-b13a)
Aug 14 13:44:07 VERBOSE[33085] chan_sip.c: Hangup call SIP/pbxes.com-b13a, SIP callid 14024154086f20d772e960ae28b673e5@144.76.38.78
Aug 14 13:44:07 VERBOSE[33085] logger.c: -- Local/holbrook-105@from-pstn/n-d4fe,1 answered SIP/holbrook-601-31d4
Aug 14 13:44:07 VERBOSE[33085] logger.c: We're at 144.76.38.78 port 42894
Aug 14 13:44:07 VERBOSE[33085] logger.c: Video is at 144.76.38.78 port 37166
Aug 14 13:44:07 VERBOSE[33085] logger.c: Adding codec 0x4 (ulaw) to SDP
Aug 14 13:44:07 VERBOSE[33085] logger.c: Adding codec 0x8 (alaw) to SDP
Aug 14 13:44:07 VERBOSE[33085] logger.c: Adding codec 0x200000 (h264) to SDP
Aug 14 13:44:07 VERBOSE[33085] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Aug 14 13:44:08 VERBOSE[22242] logger.c: Found RTP audio format 0
Aug 14 13:44:08 VERBOSE[22242] logger.c: Found RTP audio format 96
Aug 14 13:44:08 VERBOSE[22242] logger.c: Found RTP video format 97
Aug 14 13:44:08 VERBOSE[22242] logger.c: Peer audio RTP is at port 192.168.8.4:4020
Aug 14 13:44:08 VERBOSE[22242] logger.c: Peer video RTP is at port 192.168.8.4:4022
Aug 14 13:44:08 VERBOSE[22242] logger.c: Found description format PCMU
Aug 14 13:44:08 VERBOSE[22242] logger.c: Found description format telephone-event
Aug 14 13:44:08 VERBOSE[22242] logger.c: Found description format H264
Aug 14 13:44:08 VERBOSE[22242] logger.c: Capabilities: us - 0x38161e (gsm|ulaw|alaw|g726|speex|ilbc|g722|h263|h263p|h264), peer - audio=0x4 (ulaw)/video=0x200000 (h264), combined - 0x200004 (ulaw|h264)
Aug 14 13:44:08 VERBOSE[22242] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Aug 14 13:44:08 VERBOSE[22242] chan_sip.c: Got a SIP re-invite for call 31c35f26-bec1-4269-ad9d-ab549a87b177
Aug 14 13:44:08 VERBOSE[22242] logger.c: We're at 144.76.38.78 port 42894
Aug 14 13:44:08 VERBOSE[22242] logger.c: Video is at 144.76.38.78 port 37166
Aug 14 13:44:08 VERBOSE[22242] logger.c: Adding codec 0x4 (ulaw) to SDP
Aug 14 13:44:08 VERBOSE[22242] logger.c: Adding codec 0x200000 (h264) to SDP
Aug 14 13:44:08 VERBOSE[22242] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Aug 14 13:44:08 VERBOSE[33085] chan_sip.c: Hangup call SIP/holbrook-601-31d4, SIP callid 31c35f26-bec1-4269-ad9d-ab549a87b177


Any ideas/suggestions please??

Thread: Problem chaining ring groups
hol

Replies: 0
Views: 941

Problem chaining ring groups 18.07.2018 18:19 Forum: Miscellaneous

Can't understand why its not working, my use case is:
- Inbound route from trunk to ring group (office SIP handsets)
- after 10 seconds if not answered divert to another ring group (mobile phones created as classic extensions)
- if not answered go to voicemail on one of the office handsets

The configuration looks as if it should work, but doesn't call the 2nd ring group. With voicemail on the office phone it goes straight to that. If I turn off vm I get 'it has not been possible to connect' message.

I can ring the 2nd ring group from an office phone at that works fine.

Any ideas?

Thread: RE: IP addresses
hol

Replies: 2
Views: 1790

RE: IP addresses 22.05.2018 13:30 Forum: Miscellaneous

Thanks for the quick response.

So I make that:
144.76.38.78
107.155.198.131
5.9.79.175
67.231.240.210

Thread: RE: IP addresses
hol

Replies: 2
Views: 1790

IP addresses 22.05.2018 12:02 Forum: Miscellaneous

hi

My trunk provider is asking for the IP address of the PBX, I assume for configuring their firewall.

I see we are on www1.pbxes.com, but is there a list of IPs I can pass on to them that will cover the normal IP and any that you would use if substituting servers in a failover situation?

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