- sr-users - kamailio.org

[Serusers] How to calculate new HASH_TABLE_SIZE value when extending parser?
by Maxim Sobolev 29 Apr '03

29 Apr '03

Hi, I am currently trying to extend ser's parser to allow it recognize User-Agent headers, but stuck into the magic HASH_TABLE_SIZE value. Is there any algorithm to calculate it when adding support for a new header type? Maybe there is some documentation, apart from comments in the code itself, on extending parser, or at least on its inner details? If so, it would be nice to obtain it if possible. Thanks! -Maxim

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[Serusers] Extending usrloc module to allow to save User-Agent values [patch for review]
by Maxim Sobolev 29 Apr '03

29 Apr '03

Folks, Attached please find a patch, which extends usrloc/registrar modules to save values from User-Agent field in REGISTER messages into the database. It would be nice to have it included into the next release. Please disregard hackish detection of User-Agent fields, I am currently working on extending parser to recognize that type of field. I'll post updated patch when finished. Also, there is a new unrelated function which can be used to retrieve full list of all currently registered users in all domains. I am using this function in new version of my nathelper modules to do periodical UDP "pinging" of all registered users to keep their NAT bindings alive. I would like to have it reviewed and included into the next release as well. Thanks! -Maxim

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Re: [Serusers] Load Balancing
by Jiri Kuthan 25 Apr '03

25 Apr '03

At 03:33 PM 4/22/2003, Alejandro Olchik wrote: >I need some help to implement load balance >when sending invites to an external sip >proxy server. > >I would like to be able to balance load >between to IPs (10.0.0.1 and 10.0.0.2) and >use the backup IP when the primary one >fails. > >Below is the routing code I have: > > > if (!lookup("location")) { > rewritehost("10.0.0.1"); > if (!t_relay_to("10.0.0.1","5060")) { > sl_send_reply("404", "Not Found"); > }; > break; > }; > >How can I add this behaviour? That depends on the distribution scheme you would like to use. For example, you can implement a distribution function that splits requests based on some downstream weights. e.g., modparam("distrib", "weights", # ... # rewrite host-part with one of values using the distribution # 20-30-50 distribute("10.0.0.1/20%, 10.0.0.2/30%, 10.0.0.3/50%) t_on_negative("1") t_relay_to(); # ... reply_route[1] { # look at destinations that failed, and update their probabilities # according to some strategy; for example, temporary 0% weight -- # note that this would take shared memory for keeping the weights # and a timer update_weights(); # if coupled to TM, it can look at previous attempts to eliminate # retrying to a previously failed destination distribute("10.0.0.1/20%, 10.0.0.2/30%, 10.0.0.3/50%) } -Jiri

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RE: [Serusers] problem with cisco 2600 to pstn
by Dan Austin 25 Apr '03

25 Apr '03

On your pots dial-peer set the codec to g711alaw, or configure your SIP clients to use that codec. Dan -----Original Message----- From: Yang Xiang [mailto:yang.xiang@iitb.fraunhofer.de] Sent: Friday, April 25, 2003 7:55 AM To: serusers(a)lists.iptel.org Subject: [Serusers] problem with cisco 2600 to pstn Hi all, I am expericing problem with the cisco 2600, which should function as the sip2pstn gateway. If I try to complete a call from a sip phone to pstn, the router says: ------------------------------------------------------------------------ 00:15:49: ISDN BR1/0: TX -> SETUP pd = 8 callref = 0x03 00:15:49: Bearer Capability i = 0x8090A2 ^^^^^^ 00:15:49: Channel ID i = 0x83 00:15:49: Progress Ind i = 0x8181 - Call not end-to-end ISDN, may have in-band info 00:15:49: Calling Party Number i = 0x80, 'yang2', Plan:Unknown, Type:Unknown 00:15:49: Called Party Number i = 0xC9, '6091574', Plan:Private, Type:Subscriber(local) 00:15:49: ISDN BR1/0: RX <- SETUP_ACK pd = 8 callref = 0x83 00:15:49: Channel ID i = 0x89 00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate now available 00:15:49: ISDN BR1/0: RX <- DISCONNECT pd = 8 callref = 0x83 00:15:49: Cause i = 0x80C1 - Bearer capability not implemented ^^^^^^^^^^^^^^^^^^^^^^^^^^ 00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate now available ------------------------------------------------------------------------ ---- ------------ Please notice the line "Bearer Capability i = 0x8090A2", the digits "80" mean that this is a ITU voice call, "90" mean circuit mode, 64 kbps and "A2" is for G.711 u-law. So if I call the router from a normal telephone, the debugging looks as follows: ------------------------------------------------------------------------ ---- ---- 01:01:58: ISDN BR1/0: RX <- SETUP pd = 8 callref = 0x01 01:01:58: Bearer Capability i = 0x8090A3 ^^^ 01:01:58: Channel ID i = 0x89 01:01:58: Calling Party Number i = 0x0181, '609157', Plan:ISDN, Type:Unknown 01:01:58: Called Party Number i = 0xC1, '20', Plan:ISDN, Type:Subscriber(local) 01:01:58: High Layer Compat i = 0x9181 01:01:58: ISDN BR1/0: Event: Received a VOICE call from 609157 on B1 at 64 Kb/s ------------------------------------------------------------------------ ---- --------- whereat the bearer capability is "0x8090A3". It means that the ISDN-switch of German Telekom uses G.711 a-law. I am afraid that is the reason why the sip-call doesn't go through. But I can't find any way to configure this. Has anybody in this mailinglist the same experience? Any hints would be very appreciated. Thanks yang _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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Re: [Serusers] problem with cisco 2600 to pstn
by Yang Xiang 25 Apr '03

25 Apr '03

Hi Jiri, it works! Thank you very much! The voice-port block is the key: ------------------------------- voice-port 1/0/0 compand-type a-law cptone DE bearer-cap Speech ------------------------------ yang ----- Original Message ----- From: "Jiri Kuthan" <jiri(a)iptel.org> To: "Yang Xiang" <yang.xiang(a)iitb.fraunhofer.de> Sent: Friday, April 25, 2003 5:21 PM Subject: Re: [Serusers] problem with cisco 2600 to pstn > try to look at our config at http://www.iptel.org/~jiri/etc/cisco/ios_2003.txt > if it helps you. I think we had the same problem, changed some settings and it > worked then. I unfornantely don't remember what it was. > > -Jiri > > At 04:54 PM 4/25/2003, you wrote: > >Hi all, > > > >I am expericing problem with the cisco 2600, which should function as the > >sip2pstn gateway. If I try to complete a call from a sip phone to pstn, the > >router says: > >------------------------------------------------------------------------ > > 00:15:49: ISDN BR1/0: TX -> SETUP pd = 8 callref = 0x03 > >00:15:49: Bearer Capability i = 0x8090A2 > > ^^^^^^ > >00:15:49: Channel ID i = 0x83 > >00:15:49: Progress Ind i = 0x8181 - Call not end-to-end ISDN, may > >have in-band info > >00:15:49: Calling Party Number i = 0x80, 'yang2', Plan:Unknown, > >Type:Unknown > >00:15:49: Called Party Number i = 0xC9, '6091574', Plan:Private, > >Type:Subscriber(local) > >00:15:49: ISDN BR1/0: RX <- SETUP_ACK pd = 8 callref = 0x83 > >00:15:49: Channel ID i = 0x89 > >00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate now > >available > >00:15:49: ISDN BR1/0: RX <- DISCONNECT pd = 8 callref = 0x83 > >00:15:49: Cause i = 0x80C1 - Bearer capability not implemented > > > >^^^^^^^^^^^^^^^^^^^^^^^^^^ > >00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate now > >available > >--------------------------------------------------------------------------- - > >------------ > > > >Please notice the line "Bearer Capability i = 0x8090A2", the digits "80" > >mean that this is a ITU voice call, "90" mean circuit mode, 64 kbps and "A2" > >is for G.711 u-law. > > > >So if I call the router from a normal telephone, the debugging looks as > >follows: > >--------------------------------------------------------------------------- - > >---- > >01:01:58: ISDN BR1/0: RX <- SETUP pd = 8 callref = 0x01 > >01:01:58: Bearer Capability i = 0x8090A3 > > ^^^ > >01:01:58: Channel ID i = 0x89 > >01:01:58: Calling Party Number i = 0x0181, '609157', Plan:ISDN, > >Type:Unknown > >01:01:58: Called Party Number i = 0xC1, '20', Plan:ISDN, > >Type:Subscriber(local) > >01:01:58: High Layer Compat i = 0x9181 > >01:01:58: ISDN BR1/0: Event: Received a VOICE call from 609157 on B1 at 64 > >Kb/s > >--------------------------------------------------------------------------- - > >--------- > > > >whereat the bearer capability is "0x8090A3". It means that the ISDN-switch > >of German Telekom uses G.711 a-law. > > > >I am afraid that is the reason why the sip-call doesn't go through. But I > >can't find any way to configure this. > > > >Has anybody in this mailinglist the same experience? > > > >Any hints would be very appreciated. > > > >Thanks > > > >yang > > > > > > > > > > > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)lists.iptel.org > >http://lists.iptel.org/mailman/listinfo/serusers > > -- > Jiri Kuthan http://iptel.org/~jiri/ > >

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[Serusers] SERAdmin program in development to benefit all SER administrators
by Xten India 25 Apr '03

25 Apr '03

Hi , Xten Networks, Inc. (www.xten.com) is pleased to announce it is making a contribution to the SER project. The Xten SER Team is comprised of 2 senior engineers and 1 project manager who are committed full-time to the development of SERAdmin.SERAdmin is a GUI interface between SIP Express Router (SER) and the SER administrator. Project location (http://developer.berlios.de/projects/seradmin/) SERAdmin provides control over many SER tasks such as: start, stop, pause, re-start, monitor, add user, edit user, etc. SERAdmin has an intuitive look and feel.SERAdmin is open source, is being developed to benefit all SER administrators, and the feature set of SERAdmin will be determined by the iptel.org SER users' group.So please communicate with the Xten SER Team, post your comments in the public forums, and make use of the Xten SER Team as they are working for the SER community. About Xten (www.xten.com) Xten Networks, Inc. is a leading provider of high-quality SIP Voice over Internet Protocol (VoIP) software. Xten provides IP Telephony products directly to end users, the Enterprise market, Next-Gen Service Providers (ITSPs & Tier 2), Wireless Internet Service Providers (WISPs), Telephone Companies (TELCOs), and Original Equipment Manufacturers (OEMs). Regards,Xten Team --------------------------------- Do you Yahoo!? The New Yahoo! Search - Faster. Easier. Bingo.

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[Serusers] problem with cisco 2600 to pstn
by Yang Xiang 25 Apr '03

25 Apr '03

Hi all, I am expericing problem with the cisco 2600, which should function as the sip2pstn gateway. If I try to complete a call from a sip phone to pstn, the router says: ------------------------------------------------------------------------ 00:15:49: ISDN BR1/0: TX -> SETUP pd = 8 callref = 0x03 00:15:49: Bearer Capability i = 0x8090A2 ^^^^^^ 00:15:49: Channel ID i = 0x83 00:15:49: Progress Ind i = 0x8181 - Call not end-to-end ISDN, may have in-band info 00:15:49: Calling Party Number i = 0x80, 'yang2', Plan:Unknown, Type:Unknown 00:15:49: Called Party Number i = 0xC9, '6091574', Plan:Private, Type:Subscriber(local) 00:15:49: ISDN BR1/0: RX <- SETUP_ACK pd = 8 callref = 0x83 00:15:49: Channel ID i = 0x89 00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate now available 00:15:49: ISDN BR1/0: RX <- DISCONNECT pd = 8 callref = 0x83 00:15:49: Cause i = 0x80C1 - Bearer capability not implemented ^^^^^^^^^^^^^^^^^^^^^^^^^^ 00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate now available ---------------------------------------------------------------------------- ------------ Please notice the line "Bearer Capability i = 0x8090A2", the digits "80" mean that this is a ITU voice call, "90" mean circuit mode, 64 kbps and "A2" is for G.711 u-law. So if I call the router from a normal telephone, the debugging looks as follows: ---------------------------------------------------------------------------- ---- 01:01:58: ISDN BR1/0: RX <- SETUP pd = 8 callref = 0x01 01:01:58: Bearer Capability i = 0x8090A3 ^^^ 01:01:58: Channel ID i = 0x89 01:01:58: Calling Party Number i = 0x0181, '609157', Plan:ISDN, Type:Unknown 01:01:58: Called Party Number i = 0xC1, '20', Plan:ISDN, Type:Subscriber(local) 01:01:58: High Layer Compat i = 0x9181 01:01:58: ISDN BR1/0: Event: Received a VOICE call from 609157 on B1 at 64 Kb/s ---------------------------------------------------------------------------- --------- whereat the bearer capability is "0x8090A3". It means that the ISDN-switch of German Telekom uses G.711 a-law. I am afraid that is the reason why the sip-call doesn't go through. But I can't find any way to configure this. Has anybody in this mailinglist the same experience? Any hints would be very appreciated. Thanks yang

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[Serusers] msg's not reaching all the way
by magnus melin 25 Apr '03

25 Apr '03

hey all! We´re struggeling with a problem since some time now. We hade set up an SER - jabber environment, the applications are running on differnt servers. SIP-client -------(sip (udp) ) --------> SER-server -----------> Jabber-server ------(xml (tcp) ) -------> Jabber-client <------------------------------ <---------- <---- <------- Now to the problem. There is no problem with registration on either sides, and one can send a msg from the SIP- client to jabber-client, but when responding from jabber to SIP, the SER sends two MESSAGE to the sip-client and get two 400 BAD REQUEST (frame 11 & 12 ) back from the client, and then the jabber-client goes "away" (frame 19) (losses the tcp-handle ?) In short: a message is sent from the sipclient to the jabberclient, the response from the jabberclient wont display on the sip-client. Though it looks like in etherreal that the msg reaches all the way to the sip-client. We have both loose_route() and record_route() (see ser.cfg_jabber attached). Thank's Best Reg Anna & Magnus Frame 7 (520 on wire, 520 captured) Frame Number: 7 Linux cooked capture Packet type: Unicast to us (0) Source: <MISSING> Protocol: IP (0x0800) Internet Protocol, Src Addr: dhcp164-devel (172.18.3.164), Dst Addr: dhcp164-devel (172.18.3.164) Version: 4 Total Length: 504 Time to live: 64 Protocol: UDP (0x11) Source: dhcp164-devel (172.18.3.164) Destination: dhcp164-devel (172.18.3.164) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Source port: 5060 (5060) Destination port: 5060 (5060) Length: 484 Checksum: 0xb971 (correct) Session Initiation Protocol Request line: MESSAGE sip:x@e-horizon.se SIP/2.0 Message Header Via: SIP/2.0/UDP 172.18.3.164:5060;branch=z9hG4bKa858.b1ed4bd6.0 To: sip:x@e-horizon.se From: <sip:b*storstark.e-horizon.se@jstorstark.e-horizon.se>;tag=949cc72a86e24d5dcb2bf51338531b31-4809 CSeq: 10 MESSAGE Call-ID: 38231eab-9474(a)172.18.3.164 Content-Length: 20 User-Agent: Sip EXpress router(0.8.11pre13 (i386/linux)) Content-Type: text/plain Contact: <sip:b*storstark.e-horizon.se@jstorstark.e-horizon.se> Session Description Protocol Invalid line: bbbbbbbbbbbbbbbbbbbb Frame 8 (674 on wire, 674 captured) Linux cooked capture Packet type: Sent by us (4) Link-layer address type: 1 Link-layer address length: 6 Protocol: IP (0x0800) Internet Protocol, Src Addr: dhcp164-devel (172.18.3.164), Dst Addr: shot.e-horizon.se (172.18.1.66) Time to live: 64 Protocol: UDP (0x11) Header checksum: 0xdb50 (correct) Source: dhcp164-devel (172.18.3.164) Destination: shot.e-horizon.se (172.18.1.66) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 7203 (7203) Source port: 5060 (5060) Destination port: 7203 (7203) Length: 638 Checksum: 0x6016 (correct) Session Initiation Protocol Request line: MESSAGE sip:172.18.1.66:7203 SIP/2.0 Message Header Record-Route: <sip:x@172.18.3.164:5060;ftag=949cc72a86e24d5dcb2bf51338531b31-4809;lr> Via: SIP/2.0/UDP 172.18.3.164:5060;branch=z9hG4bK8984.7c66ebe.0 Via: SIP/2.0/UDP 172.18.3.164:5060;branch=z9hG4bKa858.b1ed4bd6.0 To: sip:x@e-horizon.se From: <sip:b*storstark.e-horizon.se@jstorstark.e-horizon.se>;tag=949cc72a86e24d5dcb2bf51338531b31-4809 CSeq: 10 MESSAGE Call-ID: 38231eab-9474(a)172.18.3.164 Content-Length: 20 User-Agent: Sip EXpress router(0.8.11pre13 (i386/linux)) Content-Type: text/plain Contact: <sip:b*storstark.e-horizon.se@jstorstark.e-horizon.se> Session Description Protocol Invalid line: bbbbbbbbbbbbbbbbbbbb Frame 9 (520 on wire, 520 captured) Linux cooked capture Source: <MISSING> Protocol: IP (0x0800) Internet Protocol, Src Addr: dhcp164-devel (172.18.3.164), Dst Addr: dhcp164-devel (172.18.3.164) Protocol: UDP (0x11) Header checksum: 0xd988 (correct) Source: dhcp164-devel (172.18.3.164) Destination: dhcp164-devel (172.18.3.164) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Source port: 5060 (5060) Destination port: 5060 (5060) Length: 484 Session Initiation Protocol Request line: MESSAGE sip:x@e-horizon.se SIP/2.0 Message Header Via: SIP/2.0/UDP 172.18.3.164:5060;branch=z9hG4bKa858.b1ed4bd6.0 To: sip:x@e-horizon.se From: <sip:b*storstark.e-horizon.se@jstorstark.e-horizon.se>;tag=949cc72a86e24d5dcb2bf51338531b31-4809 CSeq: 10 MESSAGE Call-ID: 38231eab-9474(a)172.18.3.164 Content-Length: 20 User-Agent: Sip EXpress router(0.8.11pre13 (i386/linux)) Content-Type: text/plain Contact: <sip:b*storstark.e-horizon.se@jstorstark.e-horizon.se> Session Description Protocol Invalid line: bbbbbbbbbbbbbbbbbbbb Frame 10 (674 on wire, 674 captured) Arrival Time: Apr 23, 2003 15:28:20.284421000 Linux cooked capture Packet type: Sent by us (4) Link-layer address type: 1 Link-layer address length: 6 Protocol: IP (0x0800) Internet Protocol, Src Addr: dhcp164-devel (172.18.3.164), Dst Addr: shot.e-horizon.se (172.18.1.66) Version: 4 Header length: 20 bytes Protocol: UDP (0x11) Header checksum: 0xdb50 (correct) Source: dhcp164-devel (172.18.3.164) Destination: shot.e-horizon.se (172.18.1.66) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 7203 (7203) Source port: 5060 (5060) Destination port: 7203 (7203) Session Initiation Protocol Request line: MESSAGE sip:172.18.1.66:7203 SIP/2.0 Message Header Record-Route: <sip:x@172.18.3.164:5060;ftag=949cc72a86e24d5dcb2bf51338531b31-4809;lr> Via: SIP/2.0/UDP 172.18.3.164:5060;branch=z9hG4bK8984.7c66ebe.0 Via: SIP/2.0/UDP 172.18.3.164:5060;branch=z9hG4bKa858.b1ed4bd6.0 To: sip:x@e-horizon.se From: <sip:b*storstark.e-horizon.se@jstorstark.e-horizon.se>;tag=949cc72a86e24d5dcb2bf51338531b31-4809 CSeq: 10 MESSAGE Call-ID: 38231eab-9474(a)172.18.3.164 Content-Length: 20 User-Agent: Sip EXpress router(0.8.11pre13 (i386/linux)) Content-Type: text/plain Contact: <sip:b*storstark.e-horizon.se@jstorstark.e-horizon.se> Session Description Protocol Invalid line: bbbbbbbbbbbbbbbbbbbb Frame 11 (474 on wire, 474 captured) Arrival Time: Apr 23, 2003 15:28:20.285190000 Linux cooked capture Packet type: Unicast to us (0) Link-layer address type: 1 Link-layer address length: 6 Source: 00:05:5d:03:03:1e (D-Link_03:03:1e) Protocol: IP (0x0800) Internet Protocol, Src Addr: shot.e-horizon.se (172.18.1.66), Dst Addr: dhcp164-devel (172.18.3.164) Time to live: 127 Protocol: UDP (0x11) Header checksum: 0x765a (correct) Source: shot.e-horizon.se (172.18.1.66) Destination: dhcp164-devel (172.18.3.164) User Datagram Protocol, Src Port: 3173 (3173), Dst Port: 5060 (5060) Source port: 3173 (3173) Destination port: 5060 (5060) Length: 438 Checksum: 0x8c6c (correct) Session Initiation Protocol Status line: SIP/2.0 400 Bad Request Message Header Via: SIP/2.0/UDP 172.18.3.164:5060;branch=z9hG4bK8984.7c66ebe.0 Via: SIP/2.0/UDP 172.18.3.164:5060;branch=z9hG4bKa858.b1ed4bd6.0 From: <sip:b*storstark.e-horizon.se@jstorstark.e-horizon.se>;tag=949cc72a86e24d5dcb2bf51338531b31-4809 To: sip:x@e-horizon.se;tag=199332f8-e402-4795-833a-c90969c979b0 Call-ID: 38231eab-9474(a)172.18.3.164 CSeq: 10 MESSAGE User-Agent: Windows RTC/1.0 Content-Length: 0 Frame 12 (409 on wire, 409 captured) Arrival Time: Apr 23, 2003 15:28:20.288277000 Linux cooked capture Packet type: Unicast to us (0) Link-layer address type: 772 Link-layer address length: 0 Source: <MISSING> Protocol: IP (0x0800) Internet Protocol, Src Addr: dhcp164-devel (172.18.3.164), Dst Addr: dhcp164-devel (172.18.3.164) Time to live: 64 Protocol: UDP (0x11) Header checksum: 0xd9f7 (correct) Source: dhcp164-devel (172.18.3.164) Destination: dhcp164-devel (172.18.3.164) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Source port: 5060 (5060) Destination port: 5060 (5060) Length: 373 Checksum: 0x4b0e (correct) Session Initiation Protocol Status line: SIP/2.0 400 Bad Request Message Header Via: SIP/2.0/UDP 172.18.3.164:5060;branch=z9hG4bKa858.b1ed4bd6.0 From: <sip:b*storstark.e-horizon.se@jstorstark.e-horizon.se>;tag=949cc72a86e24d5dcb2bf51338531b31-4809 To: sip:x@e-horizon.se;tag=199332f8-e402-4795-833a-c90969c979b0 Call-ID: 38231eab-9474(a)172.18.3.164 CSeq: 10 MESSAGE User-Agent: Windows RTC/1.0 Content-Length: 0 . . . . Frame 19 (530 on wire, 530 captured) Frame Number: 19 Internet Protocol, Src Addr: dhcp164-devel (172.18.3.164), Dst Addr: dhcp164-devel (172.18.3.164) Version: 4 Header length: 20 bytes Time to live: 64 Protocol: UDP (0x11) Source: dhcp164-devel (172.18.3.164) Destination: dhcp164-devel (172.18.3.164) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Source port: 5060 (5060) Destination port: 5060 (5060) Session Initiation Protocol Request line: MESSAGE sip:x@e-horizon.se SIP/2.0 Message Header Via: SIP/2.0/UDP 172.18.3.164:5060;branch=z9hG4bKb649.52575b6.0 To: sip:x@e-horizon.se From: <sip:sip_to_jabber_gateway>;tag=949cc72a86e24d5dcb2bf51338531b31-dc30 CSeq: 10 MESSAGE Call-ID: 38231eac-9474(a)172.18.3.164 Content-Length: 85 User-Agent: Sip EXpress router(0.8.11pre13 (i386/linux)) Content-Type: text/plain Contact: <sip:sip_to_jabber_gateway> Session Description Protocol Invalid line: INFO: Your are now offline in Jabber network. Thank you for using SIP-Jabber gateway.

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[Serusers] serweb & usrloc
by kelch 25 Apr '03

25 Apr '03

hi! is serweb intended to just send IM one-way? i configured everything and noticed that i can only send messages from a web-client (via serweb) to a sip softphone, but not the other way around. i noticed that this is due to the fact that when a user logs on to serweb, it does not put anything in the location table in ser db. so if a message was sent intended for that user, the sender receives 404 not found (due to failure in lookup()). what i am thinking as a workaround is to manipulate the php files in serweb to insert the user into the 'location' table via fifo, the method though crude might just work... my problem is, before i could try to do the workaround is whether serweb would accept the message if it indeed received one on port 5060? would it? or should the proxy send the message through another port like 80?

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[Serusers] MOTOROLA T720i ��:229$
by 236553＠delphi.com 25 Apr '03

25 Apr '03

1 0

Re: [Serusers] Testing Concurrent Calls
by Jiri Kuthan 24 Apr '03

24 Apr '03

I'm not aware of an affordable stress tools which covers both media and signaling. -Jiri At 03:50 PM 4/24/2003, Alejandro Olchik wrote: >I would like to start multiple simultaneous calls >(let's say around 30), and keep calls active to >test concurrency. > >Any suggestion how can I do that without requiring >expensive software for that? > >I have enough ports to terminate the calls using >G729 but I don't know what to use for originate them. > >Regards, >Alejandro > > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] how configure ser for pstn?
by Yang Xiang 24 Apr '03

24 Apr '03

Hi all, I put following config in the main routing-block for forwarding INVITE-msg to pstn-gatway: ... route{ ... if (!t_relay()) { sl_reply_error(); }; # forward message to PSTN Gateway if (uri=~"^sip:[0-9]*@"){ log(5, "Forward to pstn \n"); forward("10.20.0.6"); break; }; } However, SER doesn't do it and sends a reply "404 not found" back instead. Is it the correct position to put the function "forward()" there? By the way, ser also doesn't write log message to syslog. Following is my syslog.conf, is anything wrong in the syslog.conf? #ident "@(#)syslog.conf 1.5 98/12/14 SMI" /* SunOS 5.0 */ # # Copyright (c) 1991-1998 by Sun Microsystems, Inc. # All rights reserved. # # syslog configuration file. # # This file is processed by m4 so be careful to quote (`') names # that match m4 reserved words. Also, within ifdef's, arguments # containing commas must be quoted. # *.debug;*.notice;*.info;*.crit;*.alert /var/log/debug *.err;kern.notice;auth.notice /dev/sysmsg *.err;kern.debug;daemon.notice;mail.crit /var/adm/messages *.alert;kern.err;daemon.err operator *.alert root *.emerg * # if a non-loghost machine chooses to have authentication messages # sent to the loghost machine, un-comment out the following line: #auth.notice ifdef(`LOGHOST', /var/log/authlog, @loghost) mail.debug ifdef(`LOGHOST', /var/log/syslog, @loghost) # # non-loghost machines will use the following lines to cause "user" # log messages to be logged locally. # user.err /dev/sysmsg user.err /var/adm/messages user.alert `root, operator' user.emerg * auth.info /var/adm/messages local6.debug /var/adm/imapd.log auth.debug /var/adm/auth.log Thanks, Yang

2 1

[Serusers] Testing Concurrent Calls
by Alejandro Olchik 24 Apr '03

24 Apr '03

I would like to start multiple simultaneous calls (let's say around 30), and keep calls active to test concurrency. Any suggestion how can I do that without requiring expensive software for that? I have enough ports to terminate the calls using G729 but I don't know what to use for originate them. Regards, Alejandro

1 0

[Serusers] notify
by Kelvin Chua 24 Apr '03

24 Apr '03

how do you send notify? sl_send_reply("200","notify")? i don't think this is correct, i can't seem to find anything about it in the docs?

2 1

[Serusers] , about shared memory
by jimmy_huang 24 Apr '03

24 Apr '03

Hi , In Ser's Makefile.defs , originally , it define PKG_MALLOC , SHM_MEM , SHM_MMAP ,for shared memory, and I would like to marked it , cause i want to try let it not use shared memory (and modified ser.cfg let it just produce one main parent process ) I tried marked those three definitions , but it has error when compile , it seems some modules still need functions which only exist when those three definitions is defined, do those three definitions can't marked? or in ser , it can't work without using shared memory , can't mark those definition? Thanks , Jimmy

3 2

Re: [Serusers] Startup
by 'Jan Janak' 24 Apr '03

24 Apr '03

On 23-04 19:35, Dinesh wrote: > > Yes I do have "fork-yes" > > In cfg > > Should I change it to no? No. Send us your start-up script. And please reply to serusers(a)lists.iptel.org, not directly to me. Jan.

3 3

[Serusers] cisco 2600 as sip-pstn gateway
by Ng, Soo Sim 23 Apr '03

23 Apr '03

This is what you need 1 NM-2V + VIC-2BRI SSng

1 0

[Serusers] cisco 2600 as sip-pstn gateway
by Yang Xiang 23 Apr '03

23 Apr '03

Hi all, I want to use a cisco 2600 as sip2pstn gateway. In csico dokumentations and in postings of other people it seems that the cisco 2600 works with a PRI ISDN interface. Does anybody konw if it also works with BRI interface? I don't find a way to define "voice-port" on the router just with BRI interface. Has anybody here successfully tested this? Many thanks Yang

2 2

RE: [Serusers] cisco 2600 as sip-pstn gateway
by Dan Austin 23 Apr '03

23 Apr '03

The 'just a BRI' is the tricky part. You should need a NM-HDV (I think that's the part number), with a 2 port BRI VWIC. Dan -----Original Message----- From: Greg Fausak [mailto:greg@august.net] Sent: Wednesday, April 23, 2003 9:51 AM To: 'Yang Xiang'; serusers(a)lists.iptel.org Subject: RE: [Serusers] cisco 2600 as sip-pstn gateway Yes, it works with a BRI interface. I have experimented with a 3620 like that, the Same cards go in the 26xx series I believe. ---greg > -----Original Message----- > From: serusers-admin(a)lists.iptel.org > [mailto:serusers-admin@lists.iptel.org] On Behalf Of Yang Xiang > Sent: Wednesday, April 23, 2003 11:32 AM > To: serusers(a)lists.iptel.org > Subject: [Serusers] cisco 2600 as sip-pstn gateway > > > Hi all, > > I want to use a cisco 2600 as sip2pstn gateway. In csico > dokumentations and > in postings of other people it seems that the cisco 2600 > works with a PRI > ISDN interface. Does anybody konw if it also works with BRI > interface? I > don't find a way to define "voice-port" on the router just with BRI > interface. Has anybody here successfully tested this? > > Many thanks > > Yang > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers > _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] fed up and need guidance
by Mat Harris 23 Apr '03

23 Apr '03

hi, i have been playing around with ser now for around a month and am getting nowhere. i have two sites running ser on redhat (7.3 and 8). the ser machines are each behind a redhat (7.3) firewall with port 5060(udp) forwarded to the ser machines. i cannot see presence of any users at the other site and vice versa. please can someone guide me through how they would set up such an operation. this really is a hand-holding job i'm afraid. very grateful for a quick response. cheers -- Mat Harris OpenGPG Public Key ID: C37D57D9 mat.harris(a)genestate.com www.genestate.com

7 19

Re: [Serusers] Groups
by 'Jan Janak' 23 Apr '03

23 Apr '03

Hello, On 23-04 19:28, Dinesh wrote: > Thanks for your comments but > > > 1. Could you confirm I would need to change the mysql db manually as > serctl restricts the group names it accepts Yes, edit serctl script and look for ACL_GROUPS variable, list the groups you want to use there. > 2. Could you give me a clue as to the script I would need to add. As I > cannot see how to check the called parties groups. is_user_in("To", "group") Jan.

1 0

[Serusers] Load Balancing
by Alejandro Olchik 22 Apr '03

22 Apr '03

I need some help to implement load balance when sending invites to an external sip proxy server. I would like to be able to balance load between to IPs (10.0.0.1 and 10.0.0.2) and use the backup IP when the primary one fails. Below is the routing code I have: if (!lookup("location")) { rewritehost("10.0.0.1"); if (!t_relay_to("10.0.0.1","5060")) { sl_send_reply("404", "Not Found"); }; break; }; How can I add this behaviour? Any suggestion? Alejandro

2 1

[Serusers] show active calls
by Bo 22 Apr '03

22 Apr '03

Is there any way to show active calls (dialed number, duration, originating IP...) with SER? Thanks.

2 2

[Serusers] RFC3261 Compliance
by Andrew Sze 22 Apr '03

22 Apr '03

Hello, I would like to know whether the current SER is RFC2543 or RFC3261 compliant. It is described to be a RFC3261 server in the webpage but is specified to be RFC2543 compliant in the product sheet. (I want to use SER to test with other RFC3261 devices.) Thank you. Andrew

2 1

Re: [Serusers] fed up and need guidance
by jaime.gill＠orange.co.uk 22 Apr '03

22 Apr '03

Hi Mat, Just saw your question posted in the mailing list. If I understand your scenario is like this: UA1 --- SER ---- NAT/FW --- Internet ---- NAT/FW ----- SER ---- UA2 If so, I have been working to get something like this working for a couple of months. I wrote a module that is supposed to act as a nathelper and as a FCP client. If you are still having problems to get this working, get back to me and I can provide you with guidance. Regards, Jaime Mat Harris <mat.harris(a)genestate.com> on 16/04/2003 14:40:35 To: serusers(a)lists.iptel.org cc: (bcc: Jaime GILL/EN/HTLUK) Subject: [Serusers] fed up and need guidance hi, i have been playing around with ser now for around a month and am getting nowhere. i have two sites running ser on redhat (7.3 and 8). the ser machines are each behind a redhat (7.3) firewall with port 5060(udp) forwarded to the ser machines. i cannot see presence of any users at the other site and vice versa. please can someone guide me through how they would set up such an operation. this really is a hand-holding job i'm afraid. very grateful for a quick response. cheers -- Mat Harris OpenGPG Public Key ID: C37D57D9 mat.harris(a)genestate.com www.genestate.com ******************************************************************************* Important. Confidentiality: This communication is intended for the above-named person and may be confidential and/or legally privileged. Any opinions expressed in this communication are not necessarily those of the company. If it has come to you in error you must take no action based on it, nor must you copy or show it to anyone; please delete/destroy and inform the sender immediately. Monitoring/Viruses Orange may monitor all incoming and outgoing emails in line with current legislation. Although we have taken steps to ensure that this email and attachments are free from any virus, we advise that in keeping with good computing practice the recipient should ensure they are actually virus free. Orange PCS Limited is a subsidiary of Orange SA and is registered in England No 2178917, with its address at St James Court, Great Park Road, Almondsbury Park, Bradley Stoke, Bristol BS32 4QJ. *******************************************************************************

1 0

[Serusers] Illegal instruction
by Matthew McGillis 21 Apr '03

21 Apr '03

System Configuration: RedHat 8.0 Linux box Linux red.jenika.com 2.4.18-27.8.0 #1 Fri Mar 14 05:45:24 EST 2003 i586 i586 i386 GNU/Linux make-3.79.1-14 tar-1.13.25-8 bison-1.35-4 gcc-3.2-7 libgcc-3.2-7 flex-2.5.4a-26 Downloaded ser-0.8.10 source. make builds everything and install installed. However this is what I always get after executing it: $ /usr/local/sbin/ser Illegal instruction This is with debug: $ /usr/local/sbin/ser -dddd -E 0(25834) WARNING: hash function optimized for 1024 entries 0(25834) WARNING: use of 65536 entries may lead to unflat distribution 0(25834) qm_malloc_init: QM_OPTIMIZE=2048, /ROUNDTO=512 0(25834) qm_malloc_init: QM_HASH_SIZE=534, qm_block size=25656 0(25834) qm_malloc_init(0x80a5700, 1048576), start=0x80a5700 0(25834) qm_malloc_init: size= 1048576, init_overhead=25704 0(25834) qm_malloc_init: QM_OPTIMIZE=2048, /ROUNDTO=512 0(25834) qm_malloc_init: QM_HASH_SIZE=534, qm_block size=25656 0(25834) qm_malloc_init(0x4015d000, 33554432), start=0x4015d000 0(25834) qm_malloc_init: size= 33554432, init_overhead=25704 0(25834) qm_malloc(0x4015d000, 4) called from mem/shm_mem.c: shm_mem_init(275) 0(25834) qm_malloc(0x4015d000, 4) returns address 0x40163450 on 0 -th hit 0(25834) shm_mem_init: success 0(25834) qm_malloc(0x4015d000, 4) called from timer.c: init_timer(52) 0(25834) qm_malloc(0x4015d000, 4) returns address 0x40163484 on 0 -th hit 0(25834) DEBUG: register_fifo_cmd: new command (print) registered 0(25834) DEBUG: register_fifo_cmd: new command (uptime) registered 0(25834) DEBUG: register_fifo_cmd: new command (version) registered 0(25834) DEBUG: register_fifo_cmd: new command (which) registered 0(25834) DEBUG: register_fifo_cmd: new command (ps) registered Illegal instruction This is with strace: $ strace /usr/local/sbin/ser execve("/usr/local/sbin/ser", ["/usr/local/sbin/ser"], [/* 24 vars */]) = 0 uname({sys="Linux", node="red.jenika.com", ...}) = 0 brk(0) = 0x81a5760 open("/etc/ld.so.preload", O_RDONLY) = -1 ENOENT (No such file or directory) open("/etc/ld.so.cache", O_RDONLY) = 3 fstat64(3, {st_mode=S_IFREG|0644, st_size=44712, ...}) = 0 old_mmap(NULL, 44712, PROT_READ, MAP_PRIVATE, 3, 0) = 0x40013000 close(3) = 0 open("/lib/libdl.so.2", O_RDONLY) = 3 read(3, "\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\270\25"..., 512) = 512 fstat64(3, {st_mode=S_IFREG|0755, st_size=12165, ...}) = 0 old_mmap(NULL, 12116, PROT_READ|PROT_EXEC, MAP_PRIVATE, 3, 0) = 0x4001e000 old_mmap(0x40020000, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED, 3, 0x1000) = 0x40020000 close(3) = 0 open("/lib/libresolv.so.2", O_RDONLY) = 3 read(3, "\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\230\'\0"..., 512) = 512 fstat64(3, {st_mode=S_IFREG|0755, st_size=69443, ...}) = 0 old_mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0x40021000 old_mmap(NULL, 69444, PROT_READ|PROT_EXEC, MAP_PRIVATE, 3, 0) = 0x40022000 old_mmap(0x40030000, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED, 3, 0xe000) = 0x40030000 old_mmap(0x40031000, 8004, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x40031000 close(3) = 0 open("/lib/libc.so.6", O_RDONLY) = 3 read(3, "\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\210U\1"..., 512) = 512 fstat64(3, {st_mode=S_IFREG|0755, st_size=1377691, ...}) = 0 old_mmap(NULL, 1219172, PROT_READ|PROT_EXEC, MAP_PRIVATE, 3, 0) = 0x40033000 old_mmap(0x40155000, 24576, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED, 3, 0x121000) = 0x40155000 old_mmap(0x4015b000, 6756, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x4015b000 close(3) = 0 munmap(0x40013000, 44712) = 0 rt_sigaction(SIGINT, {0x80587b0, [INT], SA_RESTORER|SA_RESTART, 0x40059cf8}, {SIG_DFL}, 8) = 0 rt_sigaction(SIGPIPE, {0x80587b0, [PIPE], SA_RESTORER|SA_RESTART, 0x40059cf8}, {SIG_DFL}, 8) = 0 rt_sigaction(SIGUSR1, {0x80587b0, [USR1], SA_RESTORER|SA_RESTART, 0x40059cf8}, {SIG_DFL}, 8) = 0 rt_sigaction(SIGCHLD, {0x80587b0, [CHLD], SA_RESTORER|SA_RESTART, 0x40059cf8}, {SIG_DFL}, 8) = 0 rt_sigaction(SIGTERM, {0x80587b0, [TERM], SA_RESTORER|SA_RESTART, 0x40059cf8}, {SIG_DFL}, 8) = 0 rt_sigaction(SIGHUP, {0x80587b0, [HUP], SA_RESTORER|SA_RESTART, 0x40059cf8}, {SIG_DFL}, 8) = 0 rt_sigaction(SIGUSR2, {0x80587b0, [USR2], SA_RESTORER|SA_RESTART, 0x40059cf8}, {SIG_DFL}, 8) = 0 brk(0) = 0x81a5760 brk(0x81a6760) = 0x81a6760 brk(0) = 0x81a6760 brk(0x81a7000) = 0x81a7000 open("/usr/local/etc/ser/ser.cfg", O_RDONLY) = 3 open("/dev/zero", O_RDWR) = 4 old_mmap(NULL, 33554432, PROT_READ|PROT_WRITE, MAP_SHARED, 4, 0) = 0x4015d000 close(4) = 0 brk(0) = 0x81a7000 brk(0x81aa000) = 0x81aa000 ioctl(3, SNDCTL_TMR_TIMEBASE, 0xbfffd2e0) = -1 ENOTTY (Inappropriate ioctl for device) --- SIGILL (Illegal instruction) --- +++ killed by SIGILL +++ This is the stack trace from gdb: $ gdb /usr/local/sbin/ser GNU gdb Red Hat Linux (5.2.1-4) Copyright 2002 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions. Type "show copying" to see the conditions. There is absolutely no warranty for GDB. Type "show warranty" for details. This GDB was configured as "i386-redhat-linux"... (gdb) run Starting program: /usr/local/sbin/ser Program received signal SIGILL, Illegal instruction. 0x0807204b in yy_get_next_buffer () (gdb) bt #0 0x0807204b in yy_get_next_buffer () #1 0x08071cf8 in yylex () #2 0x080750ce in yyparse () #3 0x0805634e in main () #4 0x400484ed in __libc_start_main () from /lib/libc.so.6 Any ideas or thoughts would be appreciated please respond to my e-mail address I'm not a member of this mail list. Thanks! Matthew McGillis

2 1

[Serusers] SIMPLE Features of ser
by Steve Lewis 21 Apr '03

21 Apr '03

Has anyone tried out the SIMPLE features of ser? Based on what I have read it appears to support contacts and presence. If so, which clients has it been tested with? Regards, Steve

2 3

RE: [Serusers] pstn connectivity
by LeeCC, Kenny 21 Apr '03

21 Apr '03

instead of t_relay_to("xxx.xxx.xxx.xxx","5060"); try rewritehostport("xxx.xxx.xxx.xxx:5060"); > -----Original Message----- > From: budi wibowo [mailto:bu1d@yahoo.com] > Sent: Friday, April 18, 2003 09:54 > To: serusers(a)lists.iptel.org > Subject: [Serusers] pstn connectivity > > > hi i'm new guy in this list > and i have a litle problem with connectivity to psnt > thru cisco as5300 > below is my ser.cfg, yyy.yyy.yyy.yyy is ip of ser and > xx.xxx.xxx.xxx is ip of the cisco gw > > if (uri=~"^sip:[0-9]*@yyy.yyy.yyy.yyy"){ > log("Forward to pstn \n"); > t_relay_to("xxx.xxx.xxx.xxx","5060"); > break; > }; > > > > and i have this on my cisco > ! > dial-peer voice 4 voip > incoming called number . > destination-pattern 2525T > session protocol sipv2 > session target sip-server > codec g723r63 > ! > ! > sip-ua > calling-info sip-to-pstn number set 113950029 > sip-server ipv4:yyy.yyy.yy.yyyy > ! > > my pproblem is.. i cant if i dial to 252544123 from my > softphone , it's not work > but when i dial with format 252544123(a)xxx.xxx.xxx.xxx > it always work > anyone have clue for my problem? > thx > > budi > > > __________________________________________________ > Do you Yahoo!? > The New Yahoo! Search - Faster. Easier. Bingo > http://search.yahoo.com > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

5 7

[Serusers] build static modules , undefined reference to 'usrloc_exports' ?
by jimmy_huang 20 Apr '03

20 Apr '03

hello... I want to build a small and workable ser (let user can sending messages , registering , talking via ser...) , and build needed modules to static modules , what modules should I build in with static modules? I tried built tm,sl,rr,maxfwd,usrloc these modules to static modules , but it has some error...in compiling sr_module.o only when build usrloc modules to static modules, does "usrloc" modules can't build to static modules? or i have missing something?? error msg as follow... sr_module.o (.text+0x5a9):In function 'register_builtin_modules': :undefined reference to 'usrloc_exports' My environment is in RedHat 9.0 Jimmy

2 1

RE: [Serusers] pstn connectivity
by Dan Austin 20 Apr '03

20 Apr '03

OK. It sounds like the Softphone is not putting the realm/domain in the URI. Which softphone are you using? -----Original Message----- From: budi wibowo [mailto:bu1d@yahoo.com] Sent: Saturday, April 19, 2003 7:29 AM To: Dan Austin Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] pstn connectivity yes i have resistered my softphone, if i connect using the wrong username or the wrong password i got message like not authorized how do i check if my softphoen is registered or not if i dial to 1136523456(a)111.111.111.111 everything is normal tia budi --- Dan Austin <Dan_Austin(a)Phoenix.com> wrote: > Is the softphone configured to register with your > SER install? > > I've verified the setting in the how-to with MSN > messenger 4.6 > and a Cisco 7940 converted to use SIP. > > If you have to add the ip address of the gw to the > request, it > would seem like your phone might not be registered. > What happens > if you send the call to 1136523456(a)111.111.111.111? > > Dan > > -----Original Message----- > From: budi wibowo [mailto:bu1d@yahoo.com] > Sent: Friday, April 18, 2003 7:53 PM > To: serusers(a)lists.iptel.org > Subject: Re: [Serusers] pstn connectivity > > > - the request sent by the softphone > - to check the request on server ..how i do that > - i do debug ccsip all on my gateway and i saw > something uncomplete for the request , like the > translation rule not work properly > but i include the ip of cisco gw eerything is normal > any clue? or any suggestion which softphone i should > use? > > > boedz > > > > > --- Nils Ohlmeier <nils(a)iptel.org> wrote: > > Sounds like you softphone do not know where to > send > > a request if you omit the > > host part. Is the request sent by the softphone if > > you omit the host part? Is > > the request received at the server? Do the server > > deliver the request if it > > received it? and so on and on and on... > > > > Greetings > > Nils > > > > On Saturday 19 April 2003 04:31, budi wibowo > wrote: > > > dear all i repeat my problem > > > - pc running ser with ip 111.111.111.111 > > > - cisco 5350 with dial-peer to sip-server with > ip > > > 222.222.222.222 > > > - remote pc running sip softphone > > > > > > > > > i have read austin how to.. regarding how to > > connect > > > to cisco gw > > > but it's not work > > > example ..i want to dial to singapore with > pattern > > 113 > > > if i dial from my softphone 1136523456 it always > > > failed, but if dial using > > 1136523456(a)222.222.222.222 > > > it always work . > > > is there anything missing on my config? > > > > > > > > > rgds > > > > > > budi > > > __________________________________________________ > Do you Yahoo!? > The New Yahoo! Search - Faster. Easier. Bingo > http://search.yahoo.com > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers __________________________________________________ Do you Yahoo!? The New Yahoo! Search - Faster. Easier. Bingo http://search.yahoo.com

3 2

RE: [Serusers] pstn connectivity
by budi wibowo 19 Apr '03

19 Apr '03

yes i have resistered my softphone, if i connect using the wrong username or the wrong password i got message like not authorized how do i check if my softphoen is registered or not if i dial to 1136523456(a)111.111.111.111 everything is normal tia budi --- Dan Austin <Dan_Austin(a)Phoenix.com> wrote: > Is the softphone configured to register with your > SER install? > > I've verified the setting in the how-to with MSN > messenger 4.6 > and a Cisco 7940 converted to use SIP. > > If you have to add the ip address of the gw to the > request, it > would seem like your phone might not be registered. > What happens > if you send the call to 1136523456(a)111.111.111.111? > > Dan > > -----Original Message----- > From: budi wibowo [mailto:bu1d@yahoo.com] > Sent: Friday, April 18, 2003 7:53 PM > To: serusers(a)lists.iptel.org > Subject: Re: [Serusers] pstn connectivity > > > - the request sent by the softphone > - to check the request on server ..how i do that > - i do debug ccsip all on my gateway and i saw > something uncomplete for the request , like the > translation rule not work properly > but i include the ip of cisco gw eerything is normal > any clue? or any suggestion which softphone i should > use? > > > boedz > > > > > --- Nils Ohlmeier <nils(a)iptel.org> wrote: > > Sounds like you softphone do not know where to > send > > a request if you omit the > > host part. Is the request sent by the softphone if > > you omit the host part? Is > > the request received at the server? Do the server > > deliver the request if it > > received it? and so on and on and on... > > > > Greetings > > Nils > > > > On Saturday 19 April 2003 04:31, budi wibowo > wrote: > > > dear all i repeat my problem > > > - pc running ser with ip 111.111.111.111 > > > - cisco 5350 with dial-peer to sip-server with > ip > > > 222.222.222.222 > > > - remote pc running sip softphone > > > > > > > > > i have read austin how to.. regarding how to > > connect > > > to cisco gw > > > but it's not work > > > example ..i want to dial to singapore with > pattern > > 113 > > > if i dial from my softphone 1136523456 it always > > > failed, but if dial using > > 1136523456(a)222.222.222.222 > > > it always work . > > > is there anything missing on my config? > > > > > > > > > rgds > > > > > > budi > > > __________________________________________________ > Do you Yahoo!? > The New Yahoo! Search - Faster. Easier. Bingo > http://search.yahoo.com > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers __________________________________________________ Do you Yahoo!? The New Yahoo! Search - Faster. Easier. Bingo http://search.yahoo.com

1 0

[Serusers] Having difficulty running Serweb
by trung 19 Apr '03

19 Apr '03

Hi there, I have been reading Austin-how-to on the serweb and the forum, I manage to get the website up and running. But when I log into it as admin. The screen doesn't look like its getting me anywhere. It stay the same and ask to enter username and password again. If anyone have this serweb running successful on Redhat 8 or give me hint to get this running, it will be an honor. Thanks in Advance, Trung

2 1

[Serusers] error in rr.so
by Dopf Walter 18 Apr '03

18 Apr '03

hi all, in the new ver. 0.8.11 i changed additional to the news: rewriteFromRoute has been replaced with strict_route() and: 0(26945) loading module /usr/lib/ser/modules/rr.so but: 0(26945) find_export: <strict_route> not found 0(26945) find_export: <strict_route> not found 0(26945) parse error (75,15-16): unknown command, missing loadmodule? what's missing ? best wdo

4 6

[Serusers] failure in auth.so in ver 0.8.11
by Dopf Walter 18 Apr '03

18 Apr '03

hi all, i don't know what's going wrong here. i made an uprade to ver 0.8.11, since that time i get allways the same failure: 0(26268) set_mod_param_regex: auth matches module auth 0(26268) set_mod_param_regex: parameter <calculate_ha1> not found in module <auth> and 0(26268) set_mod_param_regex: auth matches module auth 0(26268) set_mod_param_regex: parameter <password_column> not found in module <auth> all commands former been handled by he module auth.so cause an error any ideas best wdo

2 1

[Serusers] pstn connectivity
by budi wibowo 18 Apr '03

18 Apr '03

hi i'm new guy in this list and i have a litle problem with connectivity to psnt thru cisco as5300 below is my ser.cfg, yyy.yyy.yyy.yyy is ip of ser and xx.xxx.xxx.xxx is ip of the cisco gw if (uri=~"^sip:[0-9]*@yyy.yyy.yyy.yyy"){ log("Forward to pstn \n"); t_relay_to("xxx.xxx.xxx.xxx","5060"); break; }; and i have this on my cisco ! dial-peer voice 4 voip incoming called number . destination-pattern 2525T session protocol sipv2 session target sip-server codec g723r63 ! ! sip-ua calling-info sip-to-pstn number set 113950029 sip-server ipv4:yyy.yyy.yy.yyyy ! my pproblem is.. i cant if i dial to 252544123 from my softphone , it's not work but when i dial with format 252544123(a)xxx.xxx.xxx.xxx it always work anyone have clue for my problem? thx budi __________________________________________________ Do you Yahoo!? The New Yahoo! Search - Faster. Easier. Bingo http://search.yahoo.com

1 0

[Serusers] Grandstream Budgettel 100 problems
by Steve Lewis 17 Apr '03

17 Apr '03

Hi All, Has anyone got a Grandstream Budgettel SIP phone to work successfully with 8.10 version of SER. When Attempting to register I get an error with what appears to be the Allow header. Actual header value is from phone is (from ethereal capture is): Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, OPTIONS Log Entry Apr 17 10:39:52 mercedes /usr/local/sbin/ser[31505]: ERROR: get_hdr_field: bad body for <Allow>(1048576) Apr 17 10:39:52 mercedes /usr/local/sbin/ser[31505]: ERROR: bad header field Apr 17 10:39:52 mercedes /usr/local/sbin/ser[31505]: ERROR: build_res_buf_from_sip_req: alas, parse_headers failed Any help would be appreciated. Steve

2 6

[Serusers] refer
by Joe_chen 17 Apr '03

17 Apr '03

hi, i want to know whether ser-0.8.10 supports the REFER method? If it does not support,when can you release the new version of ser? thank you.

2 1

[Serusers] ENUM support
by Steve Lewis 17 Apr '03

17 Apr '03

I'm interesting in experimenting with ENUM in conjunction with Ser and was wondering if there is support for enum within ser. Additionally, if anyone has experience and/or information on setting up a dns server for use with ENUM that would be appreciated. Steve Lewis

7 10

[Serusers] CPL composer
by paoasp＠libero.it 16 Apr '03

16 Apr '03

Hi all I'm an Italian Student and I'm working on CPL module. I've seen on the IPTEL site a WEB page with the CPL composer. Where could I download CPL composer software for the creation of the CPL scripts? Thanks a lot Paolo

2 1

[Serusers] linux hang
by Geetha Shree 16 Apr '03

16 Apr '03

HI, i am using redhat with kernel 2.4.18-3 and grub loader 0.91. while using SIP server, to get a detailed error log i changed some options in serv.conf file as given by server user guide. ------------------------------------- ser logs -- with default settings few logs are printed to syslog facility which typically dumps them to /var/log/messages. To enable detailed logs dumped to stderr, apply the following configuration options: debug=8, log_stderror=yes, fork=no. --------------------------------------- but after i restarted my machine it is getting hanged at boot time as it tries to start ser: while booting up somewhere at the end of booting process. and stops there. probably waiting for some response and because of that stopped system from booting normally. any idea why it is so . regards, Geetha -- ******************************************************************** eRiva provides IT Solutions & Services to companies worldwide. Our State of the Art Research & Development Centers provides Innovative Solutions to global customers. ******************************************************************** --

2 1

[Serusers] (no subject)
by anna 16 Apr '03

16 Apr '03

Hello all We have set up an environment that includes one jabberclient and one sip (msn) client. Now, the problem. Registration is no problem, everything is working fine and we se the "online" on both sides. The thing now is that our sip-client can send one or several mesages to the jabbberclient but as soon as the jabberclient send a respond, the connection is broken. When trying to send a msg from sip to jabber the sip-client just says: "The following message could not be delivered to all recipients:" Etherreal says nothing is happening, the client doesn't even try to send something out, not even to DNS or Ser. jabber to sip is no problem, and sip to jabber is no problem as long as you haven't send a msg from jabber to sip. (puh) Do you have any idea what is happening here ? Does it have to do with call-id's or the contact-field's ???? version: ser 0.8.11pre8-mem (i386/linux) Thank's in advance Regards Anna __________________________________________________________________ anna rahm ICQ#: 176699319 Current ICQ status: + More ways to contact me __________________________________________________________________

2 1

[Serusers] Improper Debian Build Depends
by Jamin W. Collins 16 Apr '03

16 Apr '03

The ser-0.8.10 sources contain a debian sub-directory for building debian packages. However the control file in this directory lists incomplete and improper build dependancies for the package. Currently the dependancies are listed as: Build-Depends: debhelper (>> 3.0.0), libmysqlclient-dev, libexpat1-dev, fakeroot While they should be: Build-Depends: debhelper (>> 3.0.0), libmysqlclient-dev, libexpat1-dev, zlib1g-dev Fakeroot is not a necessary package for building the Debian ser package. It simply makes it easier to build the package as a non-root user. -- Jamin W. Collins

3 4

[Serusers] You wrote a letter to me. Who are you?
by Luchya_Dobrynina 16 Apr '03

16 Apr '03

Hello! You have received a postcard. You can see it here: http://postcards.rin.ru/postcards/post/humour1.html Only for adults: SEX-TEST - http://eros.rin.ru/index_e.html and WALLPAPERS - http://wallpapers.rin.ru/index_e.html For music-lovers: Free 100 000 mp3-files - http://mp3.rin.ru/index_e.html And for all: Surprise! - http://humor.rin.ru/photo_e.html Goodby, Gerry. You can refuse to receive letters. For that just send a letter to unsubscribe14.com with the subject: DELETE.

1 0

[Serusers] Serious defficiencies of the t_on_negative feature
by Maxim Sobolev 15 Apr '03

15 Apr '03

Folks, I am currently playing with t_on_negative feature trying to implement overflow routing (i.e. if original destination returns an error, then request should be adjusted somehow and redirected to some, possibly different, destination). I've started with the following config: route { [...] rewriteFromRoute(); if (method == "INVITE") { addRecordRoute(); }; if (method == "INVITE) { t_on_negative("1"); }; t_relay_to("address1", "port1"); } reply_route[1] { rewritehostport("address2:port2"); append_branch(); } But quickly found that after transaction was redirected to address2:port2, ACKs, BYEs, 200K and CANCELS are still being forwarded to address1:port1, despite containing valid Route fields pointing to address2:port2. Then I've modified it as follows to let ser use information from that field to route ACKs, 200OKs and BYEs: route { [...] rewriteFromRoute(); if (method == "INVITE") { addRecordRoute(); }; if (method == "INVITE" || method == "CANCEL") { # INVITEs and CANCELs if (method == "INVITE) { t_on_negative("1"); }; t_relay_to("address1", "port1"); } else { # ACKs, 200OKs, BYEs t_relay(); }; } reply_route[1] { rewritehostport("address2:port2"); append_branch(); } For the most of the time everything works like a charm - if address1:port1 is unreachable or replies with an error the request is being redirected to the second destination, BUT if the initiating UA tries to cancel transaction when transaction is already redirected but before receiving final "200 OK" from the second destination, the CANCEL request is forwarded to address1:port1, not to address2:port2 as it should be. I've tried to modify setup as follows, thinking that maybe in the case of CANCEL explicit specification of proxy address confuses ser, but no avail - in this case ser forwards the CANCEL request to its own address and eventually it dies with Too Many Hops. route { [...] rewriteFromRoute(); if (method == "INVITE") { addRecordRoute(); }; if (method == "INVITE") { # INVITEs t_on_negative("1"); t_relay_to("address1", "port1"); } else { # CANCELs, ACKs, 200OKs, BYEs t_relay(); }; } reply_route[1] { rewritehostport("address2:port2"); append_branch(); } I think that such behaviour arises from the fact that ser after branching a transaction doesn't keep an address this transaction was forwarded to. In my opinion, it needs to be corrected, so that after receiving CANCEL from the UA that initiated transaction ser probably should CANCEL *all* branches of this transaction. To do this it needs to be able to tell exactly where to send CANCELs. What do you think? -Maxim

2 2

FWD: Re: [Serusers] Regarding SUBSCRIBE method
by Geetha Shree 15 Apr '03

15 Apr '03

---------- Original Message ---------------------------------- From: "Geetha Shree" <geethas(a)erivasystems.com> Reply-To: <geethas(a)erivasystems.com> Date: Tue, 15 Apr 2003 01:15:24 -0700 hi Thanks for the answer. But again sorry ,i am geeting 408 request time out again. I have changed the via-stack as was in the request like below >>SUBSCRIBE sip:geetha@192.168.1.9 SIP/2.0 >>Via: SIP/2.0/UDP 192.168.1.9:5060 >>Via: sip/2.0/UDP 192.168.1.22:5060 >>From: <sip:client1@192.168.1.9>;tag=1050301248710 >>To: <sip:client2@192.168.1.9> >>Call-Id: 237f8aec0b47b309dfe8af3d844f9e95(a)192.168.1.22 >>CSeq: 1 SUBSCRIBE >>Contact: <sip:192.168.1.22:5060> >>Expires: 1800 >>Content-Length: 0 >> >> >>SIP/2.0 200 OK >>Via: SIP/2.0/UDP 192.168.1.9:5060 >>Via: SIP/2.0/UDP 192.168.1.22:5060 >>From: <sip:client1@192.168.1.9>;tag=1050301248710 >>To: <sip:client2@192.168.1.9>;tag=1050301221190 >>Call-Id: 237f8aec0b47b309dfe8af3d844f9e95(a)192.168.1.22 >>CSeq: 1 SUBSCRIBE >>Contact: <sip:192.168.1.21:5060> >>Expires: 1800 >>Content-Length: 0 >the reply does not fit the request, since it has a different via stack. >It's thus not recognized as a part of the transaction by proxy, forwarded >statelessly and terminated with 408. > >-Jiri > >At 09:03 AM 4/15/2003, Geetha Shree wrote: >>hi all, >> >>We are getting 408 request timeout for our SUBSCRIBE method inspite of other Useragent sending >>a 200OK response for the SUBSCRIBE method. >> >>Both SUBSCRIBE and 200OK packets are getting proxied to respective user agents correctly. >>But the client1 is receiving 408 request time out error and the client2 is receiving SUBSCRIBE methods often. >> >>The SUBSCRIBE packet and the 200OK packet are as shown below. >> >>SUBSCRIBE sip:geetha@192.168.1.9 SIP/2.0 >>Via: sip/2.0/UDP 192.168.1.22:5060 >>From: <sip:client1@192.168.1.9>;tag=1050301248710 >>To: <sip:client2@192.168.1.9> >>Call-Id: 237f8aec0b47b309dfe8af3d844f9e95(a)192.168.1.22 >>CSeq: 1 SUBSCRIBE >>Contact: <sip:192.168.1.22:5060> >>Expires: 1800 >>Content-Length: 0 >> >> >>SIP/2.0 200 OK >>Via: SIP/2.0/UDP 192.168.1.9:5060 >>Via: SIP/2.0/UDP 192.168.1.22:5060;received=192.168.1.22 >>From: <sip:client1@192.168.1.9>;tag=1050301248710 >>To: <sip:client2@192.168.1.9>;tag=1050301221190 >>Call-Id: 237f8aec0b47b309dfe8af3d844f9e95(a)192.168.1.22 >>CSeq: 1 SUBSCRIBE >>Contact: <sip:192.168.1.21:5060> >>Expires: 1800 >>Content-Length: 0 >> >> >> >> >>Thanks >>Geetha >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers > >-- >Jiri Kuthan http://iptel.org/~jiri/ > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers > -- ******************************************************************** eRiva provides IT Solutions & Services to companies worldwide. Our State of the Art Research & Development Centers provides Innovative Solutions to global customers. ******************************************************************** --

1 0

[Serusers] Regarding SUBSCRIBE method
by Geetha Shree 15 Apr '03

15 Apr '03

hi all, We are getting 408 request timeout for our SUBSCRIBE method inspite of other Useragent sending a 200OK response for the SUBSCRIBE method. Both SUBSCRIBE and 200OK packets are getting proxied to respective user agents correctly. But the client1 is receiving 408 request time out error and the client2 is receiving SUBSCRIBE methods often. The SUBSCRIBE packet and the 200OK packet are as shown below. SUBSCRIBE sip:geetha@192.168.1.9 SIP/2.0 Via: sip/2.0/UDP 192.168.1.22:5060 From: <sip:client1@192.168.1.9>;tag=1050301248710 To: <sip:client2@192.168.1.9> Call-Id: 237f8aec0b47b309dfe8af3d844f9e95(a)192.168.1.22 CSeq: 1 SUBSCRIBE Contact: <sip:192.168.1.22:5060> Expires: 1800 Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.9:5060 Via: SIP/2.0/UDP 192.168.1.22:5060;received=192.168.1.22 From: <sip:client1@192.168.1.9>;tag=1050301248710 To: <sip:client2@192.168.1.9>;tag=1050301221190 Call-Id: 237f8aec0b47b309dfe8af3d844f9e95(a)192.168.1.22 CSeq: 1 SUBSCRIBE Contact: <sip:192.168.1.21:5060> Expires: 1800 Content-Length: 0 Thanks Geetha

2 1

[Serusers] loose routing & M$ & Cisco
by Jiri Kuthan 14 Apr '03

14 Apr '03

FYI We found some SIP widely used implementations don't like loose-routing parameter (";lr") without any value (which is the currently documented use of loose-routing) and break. We will probably introduce a workaround option which will allow to use lr with value (e.g., ";lr=true"). In particular, we learned that Windows Messenger, rejects loose-record-routed requests and replies with "400 BAD Request" to INVITES with ";lr" in it. Cisco IOS strips all RR parameters without value away, including ";lr" from Route header fields in subsequent requests. -Jiri -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[Serusers] Problem with Presence Agent Module
by 2 2 14 Apr '03

14 Apr '03

Hi all, I downloaded the latest code from CVS and trying to include Presence Agent(pa.so) in my ser.cfg file. While trying to start the SER, i am geting "Segmentation fault" error. I have included dependent modules (tm,usrloc,jabber) before loading pa.so inside ser.cfg. Is there any special parameters to be passed for pa.so module through modparam ? Also is there a client test program available to test Presence module ? Thanks, Santosh __________________________________________________ Do you Yahoo!? Yahoo! Tax Center - File online, calculators, forms, and more http://tax.yahoo.com

3 2

[Serusers] Redhat 7.3 Compiler Question pertaining to SER
by Scott Holben 14 Apr '03

14 Apr '03

Hi, I got Pingtel, SipTone hardphones and Windows Messenger softphone to work just fine with the SER 0.8.9 server on the Internet via a Intertex IX66 firewall. I tweaked the SIP phone configurations to use the SER 0.8.10 I installed at home, from the binary RPM obtained from ftp://ftp.berlios.de/pub/ser/0.8.10/packages/redhat/7.x. I ran into some very strange behaviors with SER and the firewall. I disconnected the firewall connection to the Internet and the SIP INVITE transactions stopped involving the firewall. This was evident from analyzing Ethereal captures. (It was if the firewall was in a promiscuous mode of operation an acted upon packets that did not apply to it.) However, I still could not get INVITE transactions to generate 2xx responses for two local phone and SER proxy configured without authentication. So, I tried compiling the source code on my Redhat 7.3 box. > gcc -v Reading specs from /usr/lib/gcc-lib/i386-redhat-linux/2.96/specs gcc version 2.96 20000731 (Red Hat Linux 7.3 2.96-110) > make config <- for SER . . . Old gcc detected (2.9x), use gcc >= 3.1 for better results make: *** No rule to make target `config'. Stop. My question is should I compile SER with the GCC 2.96 compiler or pay attention to the make config message and update GCC? (Some of the RPM binaries differ from what I compiled with the default RH 7.3 GCC 2.96 compiler.) Sincerely, Scott Holben (skaht(a)iptel.org)

2 1

[Serusers] Problem with Ser configuration
by 2 2 12 Apr '03

12 Apr '03

Hi all, I just installed MySql server on my linux box and trying to start SER. I am geting an error message while staring SER: "connect_db(): Cann't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2)" My MySql serevr is runing and I can connect to SER database from my machine. However the socket is created at /tmp/mysql.sock. Now my question is, from where did SER pickup the socket path while starting ? How can I change it to /tmp/mysql.sock ? Any pointer is appreciated. Thanks and Regards, Santosh __________________________________________________ Do you Yahoo!? Yahoo! Tax Center - File online, calculators, forms, and more http://tax.yahoo.com

2 1

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