Hi, I configured Buddy Watch function on my Polycom IP 601. It works well, until I make a reload of Asterisk. After reload, It can't monitor any lines and I have to restart the phone to reactivate this function. Is this a specific problem of asterisk-1.2.3? How can I solve it? Thank in advance, regards, Marco.
We have the same issue happened to all Asterisk versions of 1.2.X (I tried all). In CLI, it shows "-- Incoming call: Got SIP response 500 "Internal Server Error" back from 192.168.2.104". Once you see this msg, the buddy watch won't work any more until you reboot the phone. I also upgrade polycom phone from version 1.6.2 to 1.6.4.0064, but no luck. Isaac -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060222/015a4411/attachment.htm
Olle E Johansson
2006-Feb-23 03:33 UTC
[Asterisk-Users] Polycom IP 601 Buddy Watch problems
Isaac Xiao wrote:> We have the same issue happened to all Asterisk versions of 1.2.X (I > tried all). In CLI, it shows ?-- Incoming call: Got SIP response 500 > "Internal Server Error" back from 192.168.2.104?. Once you see this msg, > the buddy watch won?t work any more until you reboot the phone. I also > upgrade polycom phone from version 1.6.2 to 1.6.4.0064, but no luck. >As always with SIP errors, it is very hard to say anything without a SIP transaction log. /O
Douglas Garstang
2006-Feb-23 14:44 UTC
[Asterisk-Users] Polycom IP 601 Buddy Watch problems
Polycom only supports Asterisk Business Edition. Does ABE even support hints/buddies? -----Original Message----- From: asterisk@anime.net [mailto:asterisk@anime.net] Sent: Thursday, February 23, 2006 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP 601 Buddy Watch problems On Thu, 23 Feb 2006, Nathan Bowyer wrote:>> It may have something to do with the watch limit on the Polycom firmware. > I have one phone that does this as well, nearly all the time. I believe its > the only one that exceeds the 6-7 buddy watch limit as well.Now that polycom is "committed to working with asterisk", is polycom going to fix the 7 buddy watch limit? -Dan _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Here is the SIP transaction log. Caller called 7176 (Cisco 7960) from outside PSTN line, 7185(polycom 601, ip: 192.168.2.104) is the phone which monitors 7176. Reliably Transmitting (no NAT) to 192.168.2.104:5060: NOTIFY sip:7185@192.168.2.66 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.66:5060;branch=z9hG4bK567ba18b From: <sip:7176@192.168.2.66>;tag=as28665c79 To: " Wang" <sip:7185@192.168.2.66>;tag=1B2B2C20-22A9C0D1 Contact: <sip:7176@192.168.2.66> Call-ID: 67050654-692770ea-e7908aab@192.168.2.104 CSeq: 107 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 349 <?xml version="1.0"?> <!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd"> <presence> <presentity uri="sip:7185@192.168.2.66;method=SUBSCRIBE" /> <atom id="7176"> <address uri="sip:7176@192.168.2.66;user=ip" priority="0.800000"> <status status="inuse" /> <msnsubstatus substatus="onthephone" /> </address> </atom> </presence> <-- SIP read from 192.168.2.104:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 192.168.2.66:5060;branch=z9hG4bK567ba18b From: <sip:7176@192.168.2.66>;tag=as28665c79 To: " Wang" <sip:7185@192.168.2.66>;tag=1B2B2C20-22A9C0D1 CSeq: 107 NOTIFY Call-ID: 67050654-692770ea-e7908aab@192.168.2.104 Contact: <sip:7185@192.168.2.104> Event: presence User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.4.0064 Content-Length: 0 <-- SIP read from 192.168.2.104:5060: SUBSCRIBE sip:7176@192.168.2.66 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.104;branch=z9hG4bKfdb7ef6c9DE13403 From: " Wang" <sip:7185@192.168.2.66>;tag=1B2B2C20-22A9C0D1 To: <sip:7176@192.168.2.66>;tag=as28665c79 CSeq: 29 SUBSCRIBE Call-ID: 67050654-692770ea-e7908aab@192.168.2.104 Contact: <sip:7185@192.168.2.104> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.4.0064 Authorization: Digest username="7185", realm="asterisk", nonce="4eb67954", uri="sip:7176@192.168.2.66", response="3d264007cfea7ea28cf53fd4f9b12417", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 Transmitting (no NAT) to 192.168.2.104:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.104;branch=z9hG4bKfdb7ef6c9DE13403;received=192.168.2.104 From: " Wang" <sip:7185@192.168.2.66>;tag=1B2B2C20-22A9C0D1 To: <sip:7176@192.168.2.66>;tag=as28665c79 Call-ID: 67050654-692770ea-e7908aab@192.168.2.104 CSeq: 29 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 3600 Contact: <sip:7176@192.168.2.66>;expires=3600 Content-Length: 0 --- Reliably Transmitting (no NAT) to 192.168.2.104:5060: NOTIFY sip:7185@192.168.2.66 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.66:5060;branch=z9hG4bK7601a9bd From: <sip:7176@192.168.2.66>;tag=as28665c79 To: " Wang" <sip:7185@192.168.2.66>;tag=1B2B2C20-22A9C0D1 Contact: <sip:7176@192.168.2.66> Call-ID: 67050654-692770ea-e7908aab@192.168.2.104 CSeq: 108 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 349 <?xml version="1.0"?> <!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd"> <presence> <presentity uri="sip:7185@192.168.2.66;method=SUBSCRIBE" /> <atom id="7176"> <address uri="sip:7176@192.168.2.66;user=ip" priority="0.800000"> <status status="inuse" /> <msnsubstatus substatus="onthephone" /> </address> </atom> </presence> <-- SIP read from 192.168.2.104:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 192.168.2.66:5060;branch=z9hG4bK7601a9bd From: <sip:7176@192.168.2.66>;tag=as28665c79 To: " Wang" <sip:7185@192.168.2.66>;tag=1B2B2C20-22A9C0D1 CSeq: 108 NOTIFY Call-ID: 67050654-692770ea-e7908aab@192.168.2.104 Contact: <sip:7185@192.168.2.104> Event: presence User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.4.0064 Content-Length: 0 Reliably Transmitting (no NAT) to 192.168.2.104:5060: NOTIFY sip:7185@192.168.2.66 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.66:5060;branch=z9hG4bK212f2520 From: <sip:7176@192.168.2.66>;tag=as28665c79 To: " Wang" <sip:7185@192.168.2.66>;tag=1B2B2C20-22A9C0D1 Contact: <sip:7176@192.168.2.66> Call-ID: 67050654-692770ea-e7908aab@192.168.2.104 CSeq: 109 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 344 <?xml version="1.0"?> <!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd"> <presence> <presentity uri="sip:7185@192.168.2.66;method=SUBSCRIBE" /> <atom id="7176"> <address uri="sip:7176@192.168.2.66;user=ip" priority="0.800000"> <status status="open" /> <msnsubstatus substatus="online" /> </address> </atom> </presence> <-- SIP read from 192.168.2.104:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 192.168.2.66:5060;branch=z9hG4bK212f2520 From: <sip:7176@192.168.2.66>;tag=as28665c79 To: " Wang" <sip:7185@192.168.2.66>;tag=1B2B2C20-22A9C0D1 CSeq: 109 NOTIFY Call-ID: 67050654-692770ea-e7908aab@192.168.2.104 Contact: <sip:7185@192.168.2.104> Event: presence User-Agent: 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