Displaying 7 results from an estimated 7 matches for "spip_501".
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spip_500
2008 Oct 01
1
No reply to our critical packet
...;IP : 74.CENSORED.213 Port 5060
Defaddr->IP : 0.0.0.0 Port 5060
Reg. exten :
Def. Username: 17865221569
SIP Options : (none)
Codecs : 0x104 (ulaw|g729)
Codec Order : (g729:20,ulaw:20)
Auto-Framing: No
Status : OK (130 ms)
Useragent : PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032
Reg. Contact : sip:17865221569 at 192.168.1.54
app5*CLI> core show version
Asterisk 1.4.21.1 built by root @ app5 on a i686 running Linux on
2008-07-09 01:41:43 UTC
2008 Nov 07
2
help with dialplan
...25AB8538-7BACFE71
To: <sip:10 at 192.168.1.8;user=phone>
CSeq: 1 INVITE
Call-ID: a4b0cbb4-9737882e-815856ff at 192.168.1.89
Contact: <sip:404 at 192.168.1.89>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.0.0258
Supported: 1?00rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 249
v=0
o=- 1226069152 1226069152 IN IP4 192.168.1.89
s=Polycom IP Phone
c=IN IP4 192.168.1.89
t=0 0
m=audio 2244 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCM...
2006 Jan 31
1
Forwarding issue.
...20.1.79:5060;branch=z9hG4bK271f9f8f
From: "asterisk" <sip:asterisk@10.20.1.79>;tag=as319b6dd5
To: <sip:1630@10.20.2.16>;tag=A368FBB2-B6EAC0CF
CSeq: 104 BYE
Call-ID: 758a7b251c545f4652227c4d74d2d12d@10.20.1.79
Contact: <sip:1630@10.20.2.16>
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041
Content-Length: 0
Aside from the MoH and the error, all procedes as it should; eventually,
the call is either answered or goes to voicemail.
Any ideas as to what I'm doing wrong?
Thanks,
-Ken
2006 Apr 21
0
HANGUPCAUSE on SIP channels
...yct-901@192.168.74.33:5060
Apr 21 12:35:18 WARNING[16430]: pbx.c:5983 pbx_builtin_setvar_helper:
chan=SIP/nyct-901-539f, name=SIPDOMAIN, value=192.168.74.254
Apr 21 12:35:18 WARNING[16430]: pbx.c:5983 pbx_builtin_setvar_helper:
chan=SIP/nyct-901-539f, name=SIPUSERAGENT,
value=PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041
Apr 21 12:35:18 WARNING[16430]: pbx.c:5983 pbx_builtin_setvar_helper:
chan=SIP/nyct-901-539f, name=SIPCALLID,
value=a6c4f7bd-bdfa66e3-c6fcf7a6@192.168.74.33
-- Executing Set("SIP/nyct-901-539f", "HANGUPCAUSE=1") in new stack
Apr 21 12:35:18 WARNING[16815]: p...
2011 Feb 23
0
SIP friend name
...t;
CSeq: 2 REGISTER
Call-ID: fbc4bb4b-91001d55-8c6da4c4 at 10.0.0.201
Contact: <sip:something_really_long_and_random at 10.0.0.200>;methods="INVITE,
ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017
Accept-Language: en
Authorization: Digest username="something_really_long_and_random",
realm="asterisk", nonce="35d57376", uri="sip:10.0.0.201:5060",
response="5a0596db3c3f1823b783ae195074cc5c", algorithm=MD5
Max-Forwards:...
2011 Jun 07
2
PRI issue its BUSY
Hi all,
I just configures my PRI and incoming calls are working fine but outside calling giving error PRI is BUSY :( any idea ? I have same setup on other box and that boxes works perfect.
-- DAHDI/i1/6463279153-2 is proceeding passing it to SIP/7328-00000002
-- DAHDI/i1/6463279153-2 is making progress passing it to SIP/7328-00000002
-- DAHDI/i1/6463279153-2 is busy
-- Hungup
2006 Nov 07
4
"Sticky" Polycom 501 keys and handset
Hi,
I've recently bought new Polycom 501 phones, upgraded to bootrom 3.2.2 and
SIP 2.0.1. I just noticed something, which I first blamed on Asterisk and
NATs (a 2 second silence at the beginning of a call). Something I've
noticed also on my old phone (which is having the same problem now, but its
also been upgraded).
My keys are sticky. Simple as that. Sometimes I press a number