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37302 discussions

[Serusers] more problems with cisco ios sip

by Juha Heinanen

it turned out that cisco ios sip implementation does not include the new route uri parameters (ftag, lr) that cvs ser adds in its record route entry in the requests (ack, bye, ...) that cisco sends within a dialog. what i heard is that it may take until the end of the year until these bugs get fixed. thus there is even more reasons to push cisco to get their acts together or find a better sip/pstn gw from some other vendor (so far i haven't heard of any). -- juha

21 years, 7 months

[Serusers] SIP Scenario Tool

by Greg Fausak

Howdy, I've been playing around with this SIP scenario tool. Maybe I'm just the last one to find out. But after ngrep'ing one too many packets this is really refreshing. It is a freeware application and it's output is either html or text. I've uploaded a couple of files to my website so you can see the call flow diagram capabilities. The source (perl) code can be had at: http://www1.cs.columbia.edu/sip/implementations.html under sip scenario. The example (real world debug) callflow can be viewed at my development website: http://stage.august.net/sip1_index.html and http://stage.august.net/sip1.html You use tcpdump (or ethereal, or whatever) to grab the Output, like on linux: tcpdump -s 0 -i eth0 'port 5060' -w /var/log/sip1.dump It has been kinda quiet on these lists. ---greg

21 years, 7 months

[Serusers] str2ip: WARNING: unexpected char

by Magnus Melin

Hallo First I want to thank you for your quick response the last time. When trying to send message from jabber server to the SIP client i get the following error from the SIP server, it looks like all the adresses are correct and the messages sent from SIP is properly delivered. 1(24392) DEBUG: mk_proxy: doing DNS lookup... 1(24392) str2ip: WARNING: unexpected char s in [sip.storstark] 1(24392) str2ip6: WARNING: unexpected char s in [sip.storstark] 1(24392) get_record: lookup(_sip._udp.sip.storstark, 33) failed 1(24392) sip_resolvehost: not SRV record found for sip.storstark, trying 'normal' lookup... 1(24392) str2ip: WARNING: unexpected char s in [sip.storstark] 1(24392) str2ip6: WARNING: unexpected char s in [sip.storstark] 1(24392) ERROR: mk_proxy: could not resolve hostname: "sip.storstark" 1(24392) qm_free(0x80aad40, 0x80b7aec), called from proxy.c: mk_proxy(225) 1(24392) qm_free: freeing block alloc'ed from proxy.c: mk_proxy(208) 1(24392) ERROR: t_relay: bad host name in URI <sip:x@sip.storstark> 1(24392) ERROR: uri2sock: Can't create a dst proxy 1(24392) ERROR: t_uac_dlg: no socket found Thanks in advance Best regards Magnus _________________________________________________________________ Hitta rätt på nätet med MSN Sök http://search.msn.se/

21 years, 7 months

[Serusers] Accounting with SER

by Sunithi JP

hi all, is there any minimal support for accounting with SER? rgds, sunithi

21 years, 7 months

[Serusers] SER has problems matching ACKs to replies

by Maxim Sobolev

Folks, I found that in the stateful mode, SER is having troubles matching ACKs to negative replies when Vovida B2BUA (1.5.0) is used as one UA, while Cisco ATA 186 as another one. Following is a dump of the session illustrating the problem. As you can see, SER keeps retransmiting 480, despite receiving properly formed ACKs. Does anybody have any ideas what the problem might be? -Maxim P.S. I am also seeing in the logs the following warnings: Apr 7 12:05:28 demo ser[86090]: WARNING: sip_msg_cloner: header body ignored: 4096 B2BUA->SER INVITE sip:16045215277@demo.portaone.com:5060;transport=udp;user=phone SIP/2.0 Via: SIP/2.0/UDP 64.180.102.72:5061 To: <sip:121@64.180.102.72;user=phone> From: <sip:380442466396@64.180.102.72:5061;user=phone>;tag=4505a91c4165b3ffb1b4df0ff79c2904 Call-ID: 3135130751(a)192.168.1.20 cisco-GUID: 200546447-2237180178-1764455078-2518034476 CSeq: 2 INVITE Max-Forwards: 10 Expires: 300 Contact: <sip:380442466396@64.180.102.72:5061;user=phone> User-Agent: PortaSIP (030401b) Content-Type: application/sdp Content-Length: 309 o=380442466396 20916528 20916528 IN IP4 192.168.1.20 s=ATA186 Call c=IN IP4 193.111.9.226 t=0 0 m=audio 14000 RTP/AVP 4 8 0 101 a=rtpmap:4 G723/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=direction:active a=oldmediaip:192.168.1.20 SER->B2BUA SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 64.180.102.72:5061 To: <sip:121@64.180.102.72;user=phone> From: <sip:380442466396@64.180.102.72:5061;user=phone>;tag=4505a91c4165b3ffb1b4df0ff79c2904 Call-ID: 3135130751(a)192.168.1.20 CSeq: 2 INVITE Server: Sip EXpress router (0.8.10 (i386/freebsd)) Content-Length: 0 SER->ATA INVITE sip:16045215277@172.17.1.127:5060;user=phone;transport=udp SIP/2.0 Record-Route: <sip:16045215277@64.180.102.72;branch=0> Via: SIP/2.0/UDP 64.180.102.72;branch=z9hG4bK56f9.3c3c97acad0bdb329284be13e26ccc54.0 Via: SIP/2.0/UDP 64.180.102.72:5061 To: <sip:121@64.180.102.72;user=phone> From: <sip:380442466396@64.180.102.72:5061;user=phone>;tag=4505a91c4165b3ffb1b4df0ff79c2904 Call-ID: 3135130751(a)192.168.1.20 cisco-GUID: 200546447-2237180178-1764455078-2518034476 CSeq: 2 INVITE Max-Forwards: 9 Expires: 300 Contact: <sip:380442466396@64.180.102.72:5061;user=phone> User-Agent: PortaSIP (030401b) Content-Type: application/sdp Content-Length: 309 o=380442466396 20916528 20916528 IN IP4 192.168.1.20 s=ATA186 Call c=IN IP4 193.111.9.226 t=0 0 m=audio 14000 RTP/AVP 4 8 0 101 a=rtpmap:4 G723/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=direction:active a=oldmediaip:192.168.1.20 ATA->SER SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.180.102.72;branch=z9hG4bK56f9.3c3c97acad0bdb329284be13e26ccc54.0 Via: SIP/2.0/UDP 64.180.102.72:5061 From: <sip:380442466396@64.180.102.72:5061;user=phone>;tag=4505a91c4165b3ffb1b4df0ff79c2904 To: <sip:121@64.180.102.72;user=phone>;tag=2698540533 Call-ID: 3135130751(a)192.168.1.20 CSeq: 2 INVITE Server: Cisco ATA 186 v2.15 ata18x (030401b) Content-Length: 0 ATA->SER SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 64.180.102.72;branch=z9hG4bK56f9.3c3c97acad0bdb329284be13e26ccc54.0 Via: SIP/2.0/UDP 64.180.102.72:5061 From: <sip:380442466396@64.180.102.72:5061;user=phone>;tag=4505a91c4165b3ffb1b4df0ff79c2904 To: <sip:121@64.180.102.72;user=phone>;tag=2698540533 Call-ID: 3135130751(a)192.168.1.20 CSeq: 2 INVITE Server: Cisco ATA 186 v2.15 ata18x (030401b) Content-Length: 201 Content-Type: application/sdp o=16045215277 1110 1110 IN IP4 172.17.1.127 s=ATA186 Call c=IN IP4 64.180.102.72 t=0 0 m=audio 10000 RTP/AVP 4 101 a=rtpmap:4 G723/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 SER->B2BUA SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 64.180.102.72:5061 From: <sip:380442466396@64.180.102.72:5061;user=phone>;tag=4505a91c4165b3ffb1b4df0ff79c2904 To: <sip:121@64.180.102.72;user=phone>;tag=2698540533 Call-ID: 3135130751(a)192.168.1.20 CSeq: 2 INVITE Server: Cisco ATA 186 v2.15 ata18x (030401b) Content-Length: 221 Content-Type: application/sdp o=16045215277 1110 1110 IN IP4 172.17.1.127 s=ATA186 Call c=IN IP4 64.180.102.72 t=0 0 m=audio 10000 RTP/AVP 4 101 a=rtpmap:4 G723/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=direction:active ATA->SER SIP/2.0 480 Temporarily Not Available Via: SIP/2.0/UDP 64.180.102.72;branch=z9hG4bK56f9.3c3c97acad0bdb329284be13e26ccc54.0 Via: SIP/2.0/UDP 64.180.102.72:5061 From: <sip:380442466396@64.180.102.72:5061;user=phone>;tag=4505a91c4165b3ffb1b4df0ff79c2904 To: <sip:121@64.180.102.72;user=phone>;tag=2698540533 Call-ID: 3135130751(a)192.168.1.20 CSeq: 2 INVITE Server: Cisco ATA 186 v2.15 ata18x (030401b) Content-Length: 0 SER->B2BUA SIP/2.0 480 Temporarily Not Available Via: SIP/2.0/UDP 64.180.102.72:5061 From: <sip:380442466396@64.180.102.72:5061;user=phone>;tag=4505a91c4165b3ffb1b4df0ff79c2904 To: <sip:121@64.180.102.72;user=phone>;tag=2698540533 Call-ID: 3135130751(a)192.168.1.20 CSeq: 2 INVITE Server: Cisco ATA 186 v2.15 ata18x (030401b) Content-Length: 0 B2BUA->SER ACK sip:16045215277@demo.portaone.com:5060;transport=udp;user=phone SIP/2.0 Via: SIP/2.0/UDP 64.180.102.72:5061 To: <sip:121@64.180.102.72;user=phone>;tag=2698540533 From: <sip:380442466396@64.180.102.72:5061;user=phone>;tag=4505a91c4165b3ffb1b4df0ff79c2904 Call-ID: 3135130751(a)192.168.1.20 CSeq: 2 ACK Max-Forwards: 70 Content-Length: 0 SER->ATA ACK sip:16045215277@172.17.1.127:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 64.180.102.72;branch=z9hG4bKbc52.5db2fe1211adfdd0d5db3db6e2295344.0 Via: SIP/2.0/UDP 64.180.102.72:5061 To: <sip:121@64.180.102.72;user=phone>;tag=2698540533 From: <sip:380442466396@64.180.102.72:5061;user=phone>;tag=4505a91c4165b3ffb1b4df0ff79c2904 Call-ID: 3135130751(a)192.168.1.20 CSeq: 2 ACK Max-Forwards: 69 Content-Length: 0 SER->B2BUA SIP/2.0 480 Temporarily Not Available Via: SIP/2.0/UDP 64.180.102.72:5061 From: <sip:380442466396@64.180.102.72:5061;user=phone>;tag=4505a91c4165b3ffb1b4df0ff79c2904 To: <sip:121@64.180.102.72;user=phone>;tag=2698540533 Call-ID: 3135130751(a)192.168.1.20 CSeq: 2 INVITE Server: Cisco ATA 186 v2.15 ata18x (030401b) Content-Length: 0 B2BUA->SER ACK sip:16045215277@demo.portaone.com:5060;transport=udp;user=phone SIP/2.0 Via: SIP/2.0/UDP 64.180.102.72:5061 To: <sip:121@64.180.102.72;user=phone>;tag=2698540533 From: <sip:380442466396@64.180.102.72:5061;user=phone>;tag=4505a91c4165b3ffb1b4df0ff79c2904 Call-ID: 3135130751(a)192.168.1.20 CSeq: 2 ACK Max-Forwards: 70 Content-Length: 0 SER->ATA ACK sip:16045215277@172.17.1.127:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 64.180.102.72;branch=z9hG4bKbc52.5db2fe1211adfdd0d5db3db6e2295344.0 Via: SIP/2.0/UDP 64.180.102.72:5061 To: <sip:121@64.180.102.72;user=phone>;tag=2698540533 From: <sip:380442466396@64.180.102.72:5061;user=phone>;tag=4505a91c4165b3ffb1b4df0ff79c2904 Call-ID: 3135130751(a)192.168.1.20 CSeq: 2 ACK Max-Forwards: 69 Content-Length: 0 SER->B2BUA SIP/2.0 480 Temporarily Not Available Via: SIP/2.0/UDP 64.180.102.72:5061 From: <sip:380442466396@64.180.102.72:5061;user=phone>;tag=4505a91c4165b3ffb1b4df0ff79c2904 To: <sip:121@64.180.102.72;user=phone>;tag=2698540533 Call-ID: 3135130751(a)192.168.1.20 CSeq: 2 INVITE Server: Cisco ATA 186 v2.15 ata18x (030401b) Content-Length: 0 B2BUA->SER ACK sip:16045215277@demo.portaone.com:5060;transport=udp;user=phone SIP/2.0 Via: SIP/2.0/UDP 64.180.102.72:5061 To: <sip:121@64.180.102.72;user=phone>;tag=2698540533 From: <sip:380442466396@64.180.102.72:5061;user=phone>;tag=4505a91c4165b3ffb1b4df0ff79c2904 Call-ID: 3135130751(a)192.168.1.20 CSeq: 2 ACK Max-Forwards: 70 Content-Length: 0 SER->ATA ACK sip:16045215277@172.17.1.127:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 64.180.102.72;branch=z9hG4bKbc52.5db2fe1211adfdd0d5db3db6e2295344.0 Via: SIP/2.0/UDP 64.180.102.72:5061 To: <sip:121@64.180.102.72;user=phone>;tag=2698540533 From: <sip:380442466396@64.180.102.72:5061;user=phone>;tag=4505a91c4165b3ffb1b4df0ff79c2904 Call-ID: 3135130751(a)192.168.1.20 CSeq: 2 ACK Max-Forwards: 69 Content-Length: 0 [the last 3 repeat until timeout hits]

21 years, 7 months

[Serusers] �� Wolkswagen Golf 4. 1999 �.�.

by 236260＠fox.iptel.org

21 years, 7 months

[Serusers] About Aliases

by Santosh M Hulkund

Hi, How do I make the aliases field editable in serweb. The user should have the option of editing his aliases. Any help in this regard, greatly appreciated. Regards, Santosh M Hulkund

21 years, 7 months

[Serusers] FCP module getting ready for testing...

by jaime＠umtstrial.co.uk

Hello, The FCP module for SER seems to be approaching to a stable version (no guarranties, though). The module is in charge of 2 functions: - Translation of SIP messages from a private network to a public one (using NAT). - Open and close pinholes in a firewall to allow media sessions through. The module implements the FCP side that communicates with a deamon called fcpd that runs on the firewall itself (developed by iptel: http://www.iptel.org/fcp/). It seems to be reasonably stable now with cvs version of ser from 31/03/03. I'm looking for anyone interested in trying it and provide any feedback (bugs, new features or complains :)). Greetings, Jaime

21 years, 7 months

[Serusers] Suppress Warning messages

by Santosh M Hulkund

Hi all, How do I suppress warning messages in serweb ? Regards, Santosh M Hulkund

21 years, 7 months

Re: [Serusers] SIP and ATA-186 forwarding

by Jiri Kuthan

At 09:34 PM 4/6/2003, Mike wrote: >I apologize for the terminology, I'm still learning. Don't worry -- actually there was some confusion in what I wrote you too. ATA may do BYE/Also, but the standard way of call-transfer is using REFER request. >Maybe not an "issue" with SER, but I guess what I need is a way for SER to >get this BYE message, see the Also: and initiate another INVITE to the >correct station, handling it like a new call? No. What you need is standardized call transfer support in your phone. -Jiri

21 years, 7 months

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